Re: [OpenSIPS-Users] clusterer compatibility between 2.4 and 3.4

[Adrian Georgescu]

Because of the database schema differences between the two versions, you will 
not be able to use the same database to run both.

One migration strategy could be:

1. Convert the configuration and migrate the database structure to the new 
version 3.4 on the current slave
2. Test the slave SIP logic using a separate IP address or port with a SIP 
client that uses the slave server address as outbound SIP Proxy. When 
everything works continue to the next step.
3. Switch over the cluster to the newly configured slave running the new 
OpenSIPS version
4. Copy the configurations from the newly promoted slave to master to the old 
master machine
5. Switch back to the old master

Adrian


> On 30. Aug 2024, at 16:54, Pyle, Jeff  wrote:
> 
> Hello,
> 
> I have an OpenSIPS 2.4 cluster I need to upgrade to 3.4. The cluster is 
> configured as an HA pair, with one active and one standby where keepalived 
> moves the IP between the two.
> 
> Can instances on 2.4 and 3.4 participate in the same cluster? I'm hoping to 
> update one "half" at a time while maintaining call processing on the side 
> that isn't being upgraded.
> 
> 
> Regards,
> Jeff
> 
> This message is subject to Fusion Connect, Inc.’s email communication policy: 
> www.fusionconnect.com/email-policy 
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Re: [OpenSIPS-Users] Implementation of RFC 8599

[Adrian Georgescu]

You must register at least once, so that the server can collect the push token. 
Then, later you just wait during the Invite for the devices to get the push 
with t_wait_for_new_branches(). When any device registers again, you will get 
notified and can add a new branch to fork to. Msilo has no role into such 
arrangement.

See an example here:

https://github.com/AGProjects/sylk-pushserver/blob/master/config/opensips.cfg

—
Adrian




> On 5 Feb 2024, at 07:15, Volkan Oransoy  wrote:
> 
> Hi Adrian
> 
> Thanks for your comment. I got your point. Can you give a hint to handle the 
> the call forking for freshly registered devices? I need to keep calls in a 
> queue until the UA registers. Is msilo right approach?
> 
> Thanks
> 
> Volkan Oransoy
> On 5 Feb 2024 at 10:02 +, Adrian Georgescu , wrote:
>> You should fork to any registered contact and send push and later fork to 
>> any new devices that registered before a predefined timeout. The only flaw 
>> with this logic is that there is jo real progress indicator in SIP, you only 
>> know a push was sent but not the SIP device status like 180 Ringing, until 
>> the app comes to the foreground. 
>> 
>> If no device is registered you should still send the push otherwise it 
>> defies the purpose.
>> 
>> —
>> Adrian
>> 
>> 
>> 
>> 
>>> On 5 Feb 2024, at 05:54, Volkan Oransoy  wrote:
>>> 
>>> Yes they are but the implementation looks for a valid registration record 
>>> to initiate pn. The route snippet at the blog post looks for the pn enabled 
>>> registrations with return code 2 then pops up the E_UL_CONTACT_REFRESH 
>>> event, and waits for a registration from the UA to proxy the call. 
>>> # do lookup with method filtering
>>> lookup("location", "m");
>>> $var(rc) = $retcode;
>>> switch ($var(rc)) {
>>> case 1:
>>> # we found at least 1 non-PN contact!
>>> $var(do_relay) = true;
>>> break;
>>> case 2:
>>> # success, but all contacts are PN-enabled, so we're
>>> # sending PNs / awaiting re-registrations from them
>>> $var(do_relay) = false;
>>> break;
>>> default:
>>> xlog("L_INFO", "DBG: no contacts found ($var(rc))\n");
>>> t_reply(404, "Not Found");
>>> exit;
>>> }
>>> ...
>>> if ($var(do_relay) && !t_relay())
>>> send_reply(500, "Internal Server Error");
>>> ...
>>> 
>>> So practically, UA should keeps its registered state active, so needs a 
>>> registration refresh in regular intervals. That is not the ideal case for a 
>>> mobile app I believe. So the main idea is to keep power consumption at 
>>> minimum. 
>>> 
>>> Thanks
>>> 
>>> Volkan Oransoy
>>> On 4 Feb 2024 at 15:01 +, Adrian Georgescu , 
>>> wrote:
>>>> Push notifications were designed exactly for the case when the UA is not 
>>>> registered.  
>>>> 
>>>> —
>>>> Adrian
>>>> 
>>>> 
>>>> 
>>>> 
>>>>> On 3 Feb 2024, at 09:19, Volkan Oransoy  wrote:
>>>>> 
>>>>> Hi there
>>>>> 
>>>>> @Ronald we use that tool for notification tests and it does the job. 
>>>>> https://github.com/flutter-webrtc/callkeep/tree/master/tools
>>>>> 
>>>>> We managed to work the setup mentioned at the official blog post but that 
>>>>> requires the UA to stay registered. This needs periodic communication 
>>>>> between the UA and the registrar proxy as the RFC describes. We want the 
>>>>> call to be initiated even if the UA is offline but has a valid token 
>>>>> stored. I think we will use the msilo method with a bunch of custom 
>>>>> configs and scripts. 
>>>>> 
>>>>> Let me know if you have any comments, tips etc. 
>>>>> 
>>>>> Have a great weekend.  
>>>>> 
>>>>> Volkan Oransoy
>>>>> On 2 Feb 2024 at 14:51 +, r...@rvgeerligs.nl, wrote:
>>>>>> Hi Volkan,
>>>>>> 
>>>>>> I implemented the scripting from 
>>>>>> sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-ii/ in 
>>>>>> opensips 3.4.0.
>>>>>> As the send pn to apple or googlei s not defined I fail to get the 
>>>>>> parameters passed to the send pn to apns script which 

Re: [OpenSIPS-Users] Implementation of RFC 8599

[Adrian Georgescu]

You should fork to any registered contact and send push and later fork to any 
new devices that registered before a predefined timeout. The only flaw with 
this logic is that there is jo real progress indicator in SIP, you only know a 
push was sent but not the SIP device status like 180 Ringing, until the app 
comes to the foreground.

If no device is registered you should still send the push otherwise it defies 
the purpose.

—
Adrian




> On 5 Feb 2024, at 05:54, Volkan Oransoy  wrote:
> 
> Yes they are but the implementation looks for a valid registration record to 
> initiate pn. The route snippet at the blog post looks for the pn enabled 
> registrations with return code 2 then pops up the E_UL_CONTACT_REFRESH event, 
> and waits for a registration from the UA to proxy the call. 
> # do lookup with method filtering
> lookup("location", "m");
> $var(rc) = $retcode;
> switch ($var(rc)) {
> case 1:
> # we found at least 1 non-PN contact!
> $var(do_relay) = true;
> break;
> case 2:
> # success, but all contacts are PN-enabled, so we're
> # sending PNs / awaiting re-registrations from them
> $var(do_relay) = false;
> break;
> default:
> xlog("L_INFO", "DBG: no contacts found ($var(rc))\n");
> t_reply(404, "Not Found");
> exit;
> }
> ...
> if ($var(do_relay) && !t_relay())
> send_reply(500, "Internal Server Error");
> ...
> 
> So practically, UA should keeps its registered state active, so needs a 
> registration refresh in regular intervals. That is not the ideal case for a 
> mobile app I believe. So the main idea is to keep power consumption at 
> minimum. 
> 
> Thanks
> 
> Volkan Oransoy
> On 4 Feb 2024 at 15:01 +, Adrian Georgescu , wrote:
>> Push notifications were designed exactly for the case when the UA is not 
>> registered. 
>> 
>> —
>> Adrian
>> 
>> 
>> 
>> 
>>> On 3 Feb 2024, at 09:19, Volkan Oransoy  wrote:
>>> 
>>> Hi there
>>> 
>>> @Ronald we use that tool for notification tests and it does the job. 
>>> https://github.com/flutter-webrtc/callkeep/tree/master/tools
>>> 
>>> We managed to work the setup mentioned at the official blog post but that 
>>> requires the UA to stay registered. This needs periodic communication 
>>> between the UA and the registrar proxy as the RFC describes. We want the 
>>> call to be initiated even if the UA is offline but has a valid token 
>>> stored. I think we will use the msilo method with a bunch of custom configs 
>>> and scripts. 
>>> 
>>> Let me know if you have any comments, tips etc. 
>>> 
>>> Have a great weekend.  
>>> 
>>> Volkan Oransoy
>>> On 2 Feb 2024 at 14:51 +, r...@rvgeerligs.nl, wrote:
>>>> Hi Volkan,
>>>> 
>>>> I implemented the scripting from 
>>>> sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-ii/ in 
>>>> opensips 3.4.0.
>>>> As the send pn to apple or googlei s not defined I fail to get the 
>>>> parameters passed to the send pn to apns script which I also have.
>>>> 
>>>> I seem to have lost the device-ID in opensips or in any INVITE somewhere 
>>>> as I do not understand the RFC certainly not towards the device_ID or 
>>>> token. Or this ID is stored on registration at the proxy?
>>>> 
>>>> srcipt to directly talk to apple:
>>>> https://medium.com/@egzon.arifi/sending-a-push-notification-to-an-ios-device-using-a-bash-script-96c056c1544c
>>>> 
>>>> Please keep me posted on what you find.
>>>> 
>>>> Regards, Ronald
>>>> 
>>>> February 2, 2024 at 6:01 AM, "johan" >>> <mailto:jo...@democon.be?to=%22johan%22%20%3Cjohan%40democon.be%3E>> wrote:
>>>> 
>>>> send a query directly to apple or to google for waking them up. I did 
>>>> something like that in an external lua script.
>>>> 
>>>> On 2/02/2024 09:49, Volkan Oransoy wrote:
>>>> Hi all,
>>>> I am working on implementing RFC 8599 and have an architectural question. 
>>>> I followed the blog post 
>>>> https://blog.opensips.org/2020/06/03/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-ii/
>>>>  and it works. Based on the rfc and the implementation of Opensips, the UA 
>>>> should have a valid registration throughout the process. When a lookup 
>>>> performed, opensips finds the pn-

Re: [OpenSIPS-Users] Implementation of RFC 8599

[Adrian Georgescu]

Push notifications were designed exactly for the case when the UA is not 
registered. 

—
Adrian




> On 3 Feb 2024, at 09:19, Volkan Oransoy  wrote:
> 
> Hi there
> 
> @Ronald we use that tool for notification tests and it does the job. 
> https://github.com/flutter-webrtc/callkeep/tree/master/tools
> 
> We managed to work the setup mentioned at the official blog post but that 
> requires the UA to stay registered. This needs periodic communication between 
> the UA and the registrar proxy as the RFC describes. We want the call to be 
> initiated even if the UA is offline but has a valid token stored. I think we 
> will use the msilo method with a bunch of custom configs and scripts. 
> 
> Let me know if you have any comments, tips etc. 
> 
> Have a great weekend.  
> 
> Volkan Oransoy
> On 2 Feb 2024 at 14:51 +, r...@rvgeerligs.nl, wrote:
>> Hi Volkan,
>> 
>> I implemented the scripting from 
>> sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-ii/ in 
>> opensips 3.4.0.
>> As the send pn to apple or googlei s not defined I fail to get the 
>> parameters passed to the send pn to apns script which I also have.
>> 
>> I seem to have lost the device-ID in opensips or in any INVITE somewhere as 
>> I do not understand the RFC certainly not towards the device_ID or token. Or 
>> this ID is stored on registration at the proxy?
>> 
>> srcipt to directly talk to apple:
>> https://medium.com/@egzon.arifi/sending-a-push-notification-to-an-ios-device-using-a-bash-script-96c056c1544c
>> 
>> Please keep me posted on what you find.
>> 
>> Regards, Ronald
>> 
>> February 2, 2024 at 6:01 AM, "johan" > > wrote:
>> 
>> send a query directly to apple or to google for waking them up. I did 
>> something like that in an external lua script.
>> 
>> On 2/02/2024 09:49, Volkan Oransoy wrote:
>> Hi all,
>> I am working on implementing RFC 8599 and have an architectural question. I 
>> followed the blog post 
>> https://blog.opensips.org/2020/06/03/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-ii/
>>  and it works. Based on the rfc and the implementation of Opensips, the UA 
>> should have a valid registration throughout the process. When a lookup 
>> performed, opensips finds the pn-enabled record, triggers the PN and after a 
>> re-registration of the UA, it proxies the call. This process works without 
>> an issue. But when we kill the UA on the client device (the client is 
>> connected via WSS), after the registration expires, the call can't be 
>> proxied since there is no registration. 
>> I want to wake up the UA even if there is no registration on the proxy and 
>> suspend the call until the UA registers to the system. Should I handle this 
>> scenario with a custom setup? How do you handle this scenario on your 
>> setups? Or do I misinterpret the rfc? 
>> Thanks
>> 
>> Volkan
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Re: [OpenSIPS-Users] pn push notifications

[Adrian Georgescu]

OpenSIPS handles the SIP traffic, of course. The push server just handles 
multiple combinations of apps/ push services in an easier way. You could do a 
lot or all that its needed just from within OpenSIPS but as the number of 
combinations of apps and services grow another layer of abstraction make it 
easier to manage, plus you eliminate another possible blocking item from your 
routes as the works are down asynchronously outside OpenSIPS.
 
—
Adrian




> On 14 Jan 2024, at 20:33, r...@rvgeerligs.nl wrote:
> 
> Thank you very much, Adrian,
> 
> I noticed it allready. It seems this is a dedicated push server.
> Does it handle the normal SIP traffic as well?
> 
> Regards,
> 
> Ronald
> January 14, 2024 at 3:13 PM, "Adrian Georgescu"  <mailto:a...@ag-projects.com?to=%22Adrian%20Georgescu%22%20%3Cag%40ag-projects.com%3E>>
>  wrote:
> 
> There are many ways of implementing push notifications with OpenSIPS. This is 
> how we implemented it.
> 
> https://github.com/AGProjects/sylk-pushserver/blob/master/config/opensips.cfg
> 
> —
> Adrian
> 
> 
> 
> 
> On 13 Jan 2024, at 17:37, r...@rvgeerligs.nl <mailto:r...@rvgeerligs.nl> 
> wrote:
> Hi All,
> I am trying to implement an opensips server 3.4 with pn support. Primarily 
> apple push notifications. Later I will try google.
> Does any one have this implemented yet?
> Where can I find an example cfg for opensips?
> Any help is appreciated.
> Regards,
> Ronald Geerligs
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Re: [OpenSIPS-Users] pn push notifications

[Adrian Georgescu]

There are many ways of implementing push notifications with OpenSIPS. This is 
how we implemented it.

https://github.com/AGProjects/sylk-pushserver/blob/master/config/opensips.cfg

—
Adrian




> On 13 Jan 2024, at 17:37, r...@rvgeerligs.nl wrote:
> 
> Hi All,
> 
> 
> I am trying to implement an opensips server 3.4 with pn support. Primarily 
> apple push notifications. Later I will try google.
> 
> Does any one have this implemented yet?
> 
> Where can I find an example cfg for opensips?
> 
> Any help is appreciated.
> 
> 
> Regards,
> 
> 
> Ronald Geerligs
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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Re: [OpenSIPS-Users] Learning about resource lists

[Adrian Georgescu]

You can try these command line scripts:
Presence
sip-publish-presence 
<https://docs-new.sipthor.net/w/sip_clients/sip_publish_presence> - PUBLISH 
presence to a Presence Agent
sip-subscribe-winfo 
<https://docs-new.sipthor.net/w/sip_clients/sip_subscribe_winfo> - SUBSCRIBE to 
the watcher list for given SIP address on the Presence Agent
sip-subcribe-presence 
<https://docs-new.sipthor.net/w/sip_clients/sip_subscribe_presence> - SUBSCRIBE 
to Presence Event for a given SIP address
sip-subscribe-rls 
<https://docs-new.sipthor.net/w/sip_clients/sip_subscribe_rls> - SUBSCRIBE for 
Presence Event to a list managed by a Resource List Server
sip-subscribe-xcap-diff 
<https://docs-new.sipthor.net/w/sip_clients/sip_subscribe_xcap_diff> - 
SUBSCRIBE for xcap-diff Event to monitor changes to XCAP documents
sip-subscribe-mwi 
<https://docs-new.sipthor.net/w/sip_clients/sip_subscribe_mwi> - SUBSCRIBE for 
Message Waiting Indicator

https://sipsimpleclient.org/testing/
Testing
sipsimpleclient.org

—
Adrian




> On 22 Nov 2023, at 11:57, Robert Dyck  wrote:
> 
> Is the RLS tutorial valid?  Do we know that there are working examples?
> 
> On Wednesday, November 22, 2023 4:01:30 A.M. PST Adrian Georgescu wrote:
>> Hi Bogdan,
>> 
>> My two cents. The reality is that adoption of XCAP is practically zero. Even
>> if you build a client, you cannot make it interoperable with another, and
>> XCAP was suppose to be interoperable. If I build a buddy list on one client
>> and I cannot load it in another client, it makes no sense.
>> 
>> I think that anyone building a SIP app that needs to store/fetch data on the
>> SIP server can better do it using PUT/GET with a JSON, for example we took
>> this path for Sylk client rather than implementing XCAP again. This is not
>> interoperable between different clients, but there is no replacement
>> standard for XCAP either and is much cheaper and more reliable to do it
>> like this.
>> 
>> As far as OpenSIPS is concerned one can probably make a new module or
>> better, modify that existing RLS module so that it can read contacts
>> directly from a database table with a schema that can be defined by the
>> user. In the end what one needs is a list of URIs and a flag to see who is
>> granted to see your presence, is a very simple database model.
>> 
>> —
>> Adrian
>> 
>>> On 22 Nov 2023, at 07:02, Bogdan-Andrei Iancu  wrote:
>>> 
>>> HI Adrian,
>>> 
>>> should we understand the everything related to xcap, like RLS, buddy list,
>>> auth, etc are dropped dead at this time? if so, are you aware of any
>>> replacement  / alternatives here ?
>>> 
>>> Thanks and regards,
>>> 
>>> Bogdan-Andrei Iancu
>>> 
>>> OpenSIPS Founder and Developer
>>> 
>>>  https://www.opensips-solutions.com <https://www.opensips-solutions.com/> 
>>> <https://www.opensips-solutions.com/>
>>>  https://www.siphub.com <https://www.siphub.com/> <https://www.siphub.com/>
>>> 
>>> On 11/20/23 11:11 PM, Adrian Georgescu wrote:
>>>> XCAP is a failure. Not that we did not try, it was a bad idea and it
>>>> failed.
>>>> 
>>>> —
>>>> Adrian
>>>> 
>>>>> On 20 Nov 2023, at 14:27, Robert Dyck >>>> <mailto:rob.d...@telus.net>>
>>>>> <mailto:rob.d...@telus.net> wrote:
>>>>> 
>>>>> The context here is subscription to presence by way of a resource list.
>>>>> The learning curve is steep. I have read the tutorial. The tutorial
>>>>> gives an example of a rls-service xml document. In the example the
>>>>> resource list is contained within the services document. Various other
>>>>> examples I have found use a separate document to hold the list. The
>>>>> services document then references the list document.
>>>>> 
>>>>> https://xcap.example.com/xcap-root/resource-lists/users/s
>>>>> ip:al...@example.com 
>>>>> <mailto:al...@example.com>/index/~~/resource-lists/list%5b@name=%22l1%22%5d>>>> esource-list> If I use an integrated server the xml documents reside in
>>>>> a local database rather than the file system. Http isn't going to work.
>>>>> How would one reference the database and table using rls-services
>>>>> document? Or is a separate resource-lists document not supported when
>>>>> using an integrated rls server?
>>>>> ___
&g

Re: [OpenSIPS-Users] Learning about resource lists

[Adrian Georgescu]

You can try these command line scripts:

https://sipsimpleclient.org/testing/


Presence
• sip-publish-presence - PUBLISH presence to a Presence Agent
• sip-subscribe-winfo - SUBSCRIBE to the watcher list for given SIP address 
on the Presence Agent
• sip-subcribe-presence - SUBSCRIBE to Presence Event for a given SIP 
address
• sip-subscribe-rls - SUBSCRIBE for Presence Event to a list managed by a 
Resource List Server
• sip-subscribe-xcap-diff - SUBSCRIBE for xcap-diff Event to monitor 
changes to XCAP documents
• sip-subscribe-mwi - SUBSCRIBE for Message Waiting Indicator


—
Adrian



> On 22 Nov 2023, at 11:57, Robert Dyck  wrote:
> 
> Is the RLS tutorial valid?  Do we know that there are working examples?



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Re: [OpenSIPS-Users] Learning about resource lists

[Adrian Georgescu]

Hi Bogdan,

My two cents. The reality is that adoption of XCAP is practically zero. Even if 
you build a client, you cannot make it interoperable with another, and XCAP was 
suppose to be interoperable. If I build a buddy list on one client and I cannot 
load it in another client, it makes no sense.

I think that anyone building a SIP app that needs to store/fetch data on the 
SIP server can better do it using PUT/GET with a JSON, for example we took this 
path for Sylk client rather than implementing XCAP again. This is not 
interoperable between different clients, but there is no replacement standard 
for XCAP either and is much cheaper and more reliable to do it like this.

As far as OpenSIPS is concerned one can probably make a new module or better, 
modify that existing RLS module so that it can read contacts directly from a 
database table with a schema that can be defined by the user. In the end what 
one needs is a list of URIs and a flag to see who is granted to see your 
presence, is a very simple database model.

—
Adrian




> On 22 Nov 2023, at 07:02, Bogdan-Andrei Iancu  wrote:
> 
> HI Adrian,
> 
> should we understand the everything related to xcap, like RLS, buddy list, 
> auth, etc are dropped dead at this time? if so, are you aware of any 
> replacement  / alternatives here ?
> 
> Thanks and regards,
>  Bogdan-Andrei Iancu
> 
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com <https://www.opensips-solutions.com/>
>   https://www.siphub.com <https://www.siphub.com/>
> On 11/20/23 11:11 PM, Adrian Georgescu wrote:
>> XCAP is a failure. Not that we did not try, it was a bad idea and it failed.
>>  
>> —
>> Adrian
>> 
>> 
>> 
>> 
>>> On 20 Nov 2023, at 14:27, Robert Dyck  
>>> <mailto:rob.d...@telus.net> wrote:
>>> 
>>> The context here is subscription to presence by way of a resource list. The 
>>> learning curve is steep. I have read the tutorial. The tutorial gives an 
>>> example of a rls-service xml document. In the example the resource list is 
>>> contained within the services document. Various other examples I have found 
>>> use a separate document to hold the list. The services document then 
>>> references the list document.
>>> 
>>> https://xcap.example.com/xcap-root/resource-lists/users/sip:al...@example.com/index/~~/resource-lists/list%5b@name=%22l1%22%5d
>>> If I use an integrated server the xml documents reside in a local database 
>>> rather than the file system. Http isn't going to work. How would one 
>>> reference the database and table using rls-services document? Or is a 
>>> separate resource-lists document not supported when using an integrated rls 
>>> server?
>>> ___
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>>> Users@lists.opensips.org <mailto:Users@lists.opensips.org>
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> 
>> 
>> 
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
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Re: [OpenSIPS-Users] Learning about resource lists

[Adrian Georgescu]

XCAP is a failure. Not that we did not try, it was a bad idea and it failed.
 
—
Adrian




> On 20 Nov 2023, at 14:27, Robert Dyck  wrote:
> 
> The context here is subscription to presence by way of a resource list. The 
> learning curve is steep. I have read the tutorial. The tutorial gives an 
> example of a rls-service xml document. In the example the resource list is 
> contained within the services document. Various other examples I have found 
> use a separate document to hold the list. The services document then 
> references the list document.
> 
> https://xcap.example.com/xcap-root/resource-lists/users/sip:al...@example.com/index/~~/resource-lists/list%5b@name=%22l1%22%5d
> If I use an integrated server the xml documents reside in a local database 
> rather than the file system. Http isn't going to work. How would one 
> reference the database and table using rls-services document? Or is a 
> separate resource-lists document not supported when using an integrated rls 
> server?
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[OpenSIPS-Users] Offline messaging server

[Adrian Georgescu]

You can now use Sylk Server as SIP messaging server, is very use to use (zero 
configuration) and should interoperate easy with any SIP client or Proxy. The 
functionality is similar to what WhatsApp and Telegram offer with the 
difference that both client and server software are fully open source and based 
on open standards.

Integration guide with OpenSIPS is available at:

https://sylkserver.com/sipwebrtc-messaging-server/

Regards,
Adrian


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Re: [OpenSIPS-Users] TLS 3.2.0 experience

[Adrian Georgescu]

Using wolfssl I experience the same behaviour:

57 live01 /usr/sbin/opensips[6932]: CRITICAL:core:io_watch_del: [TCP_main] 
invalid fd -841671744, not in [0, 53)
Aug 16 20:29:57 live01 /usr/sbin/opensips[6932]: CRITICAL:core:io_watch_del: 
[TCP_main] invalid fd -841671744, not in [0, 53)
Aug 16 20:29:57 live01 /usr/sbin/opensips[6932]: CRITICAL:core:io_watch_del: 
[TCP_main] invalid fd -841671744, not in [0, 53)
Aug 16 20:29:57 live01 /usr/sbin/opensips[6932]: CRITICAL:core:io_watch_del: 
[TCP_main] invalid fd -841671744, not in [0, 53)
Aug 16 20:29:57 live01 /usr/sbin/opensips[6932]: CRITICAL:core:io_watch_del: 
[TCP_main] invalid fd -841671744, not in [0, 53)
Aug 16 20:29:57 live01 /usr/sbin/opensips[6932]: CRITICAL:core:io_watch_del: 
[TCP_main] invalid fd -841671744, not in [0, 53)
Aug 16 20:29:57 live01 /usr/sbin/opensips[6932]: CRITICAL:core:io_watch_del: 
[TCP_main] invalid fd -841671744, not in [0, 53)
Aug 16 20:29:57 live01 /usr/sbin/opensips[6932]: CRITICAL:core:io_watch_del: 
[TCP_main] invalid fd -841671744, not in [0, 53)
Aug 16 20:29:57 live01 /usr/sbin/opensips[6932]: CRITICAL:core:io_watch_del: 
[TCP_main] invalid fd -841671744, not in [0, 53)


> On 16 Aug 2021, at 15:25, Adrian Georgescu  wrote:
> 
> Also the server is using 100% CPU.
> 


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Re: [OpenSIPS-Users] auth_db module in 3.2.2

[Adrian Georgescu]

I confirm this patch fixed the issue. Thank you Liviu!

Regards,
Adrian


> On 16 Aug 2021, at 12:43, Liviu Chircu  wrote:
> 
> On 12.08.2021 22:04, Adrian Georgescu wrote:
>> Aug 12 20:51:59 live01 /usr/sbin/opensips[10064]: 
>> ERROR:db_mysql:db_mysql_store_result: driver error: Commands out of sync; 
>> you can't run this command now
>> Aug 12 20:51:59 live01 /usr/sbin/opensips[10064]: ERROR:auth_db:get_ha1: 
>> failed to query database
>> Aug 12 20:52:00 live01 /usr/sbin/opensips[10057]: 
>> ERROR:db_mysql:db_mysql_store_result: driver error: Commands out of sync; 
>> you can't run this command now
>> Aug 12 20:52:00 live01 /usr/sbin/opensips[10057]: ERROR:auth_db:get_ha1: 
>> failed to query database
> 
> Hi Adrian,
> 
> This issue should now be fixed on latest 3.2, per [1].
> 
> [1]: https://github.com/OpenSIPS/opensips/commit/c871d9edfce
> 
> Best,
> 
> -- 
> Liviu Chircu
> www.twitter.com/liviuchircu | www.opensips-solutions.com
> OpenSIPS Summit 2021 Distributed | www.opensips.org/events
> 
> 
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Re: [OpenSIPS-Users] auth_db module in 3.2.2

[Adrian Georgescu]

There are some leftovers in the module documentation related to 
password_column_2:

https://opensips.org/html/docs/modules/3.2.x/auth_db.html#param_calculate_ha1 
<https://opensips.org/html/docs/modules/3.2.x/auth_db.html#param_calculate_ha1>


The “password_column_2” column contains also HA1 strings but they should be 
calculated including the domain in the username parameter (as opposed to 
password_column which (when containing HA1 strings) should always contain HA1 
strings calculated without domain in username.


> On 16 Aug 2021, at 14:33, Adrian Georgescu  wrote:
> 
> Hi Liviu,
> 
> I understand now better the purpose of that field and is indeed not needed 
> anymore!
> 
> Regards,
> Adrian
> 
>> On 16 Aug 2021, at 12:55, Liviu Chircu  wrote:
>> 
>> On 13.08.2021 14:12, Adrian Georgescu wrote:
>>> 
>>> I would very much like to see this feature ported back to 3.2 please!
>> 
>> Hi,
>> 
>> Could you offer a bit more info on why you would want it?  Like what kind of 
>> SIP phones are still out there that cannot handle the SIP auth specs in 
>> 2021?  If you think about it, they won't work with other SIP servers: no 
>> commercial SIP server and no Asterisk, FS, PJSIP, SIP.js, drachtio, 
>> reSIPprocate, etc.:  Not a single one of these implement this crazy "ha1b" 
>> feature, and none of them give a single damn if the phone is poorly 
>> implemented and appends the "@realm" part in the username component: they 
>> will happily reply with 401 Unauthorized until the implementor fixes the 
>> phone.
>> 
>> So why should OpenSIPS have this feature? Also, I suggest you open a GitHub 
>> feature request [1] as well on this topic -- maybe we get more opinions from 
>> there as well.
>> 
>> [1]: https://github.com/OpenSIPS/opensips/issues
>> 
>> Best,
>> 
>> -- 
>> Liviu Chircu
>> www.twitter.com/liviuchircu | www.opensips-solutions.com
>> OpenSIPS Summit 2021 Distributed | www.opensips.org/events
>> 
> 

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Re: [OpenSIPS-Users] auth_db module in 3.2.2

[Adrian Georgescu]

Hi Liviu,

I understand now better the purpose of that field and is indeed not needed 
anymore!

Regards,
Adrian

> On 16 Aug 2021, at 12:55, Liviu Chircu  wrote:
> 
> On 13.08.2021 14:12, Adrian Georgescu wrote:
>> 
>> I would very much like to see this feature ported back to 3.2 please!
> 
> Hi,
> 
> Could you offer a bit more info on why you would want it?  Like what kind of 
> SIP phones are still out there that cannot handle the SIP auth specs in 2021? 
>  If you think about it, they won't work with other SIP servers: no commercial 
> SIP server and no Asterisk, FS, PJSIP, SIP.js, drachtio, reSIPprocate, etc.:  
> Not a single one of these implement this crazy "ha1b" feature, and none of 
> them give a single damn if the phone is poorly implemented and appends the 
> "@realm" part in the username component: they will happily reply with 401 
> Unauthorized until the implementor fixes the phone.
> 
> So why should OpenSIPS have this feature? Also, I suggest you open a GitHub 
> feature request [1] as well on this topic -- maybe we get more opinions from 
> there as well.
> 
> [1]: https://github.com/OpenSIPS/opensips/issues
> 
> Best,
> 
> -- 
> Liviu Chircu
> www.twitter.com/liviuchircu | www.opensips-solutions.com
> OpenSIPS Summit 2021 Distributed | www.opensips.org/events
> 


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Re: [OpenSIPS-Users] auth_db module in 3.2.2

[Adrian Georgescu]

On 12 Aug 2021, at 13:04, Liviu Chircu  wrote:
> 
> On 12.08.2021 18:36, Adrian Georgescu wrote:
>> The auth_db module has some dramatic changes which are either undocumented 
>> or not backwards compatible and is unclear how to handle this.
>> 
>> https://opensips.org/docs/modules/3.1.x/auth_db.html#param_password_column_2 
>> <https://opensips.org/docs/modules/3.1.x/auth_db.html#param_password_column_2>Hi
>>  Adrian,
> 
> Indeed, with the addition of RFC 8760 support (support for SHA-256 and 
> SHA-512-256 auth algorithms), me and Maksym Sobolyev decided to try and 
> remove the "ha1b" feature, originally designed to accommodate some broken SIP 
> UAs who cannot follow the basic SIP authentication spec.  The feature had 
> been in there since the very beginnings, and we were not sure if anyone is 
> really benefiting from it anymore nowadays.
> 
> A strong reason for removing "ha1b" was the sheer number of hashes to be 
> stored per subscriber.  Since we now have 3 algorithms (MD5, SHA-256, 
> SHA-512-256), there are 3 hash-columns to store.  With the "ha1b" feature, 
> there would be 2 x 3 = 6 hashes in total to store, per user.  So you can see 
> where this is going: "Can we get away with dropping ha1b and storing half the 
> data per user?" ... was the big question.
> 
> Still, we agreed that if there is still enough traction for the "ha1b" 
> feature from the community, we can easily re-add the ha1b logic and 3 more 
> columns to the table and backport everything to 3.2.  It's a trivial task, 
> frankly.
> 
Hi Liviu,

I would very much like to see this feature ported back to 3.2 please!

Regards,
Adrian
 


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Re: [OpenSIPS-Users] 3.2.0 TLS MGM module does not load 3.1.0 domain configuration

[Adrian Georgescu]

Thank you Vlad,

I confirm that I was able to load the certificate using wolfssl by setting the 
protocol version to TLSv1. 

Regards,
Adrian


> On 12 Aug 2021, at 18:12, Vlad Patrascu  wrote:
> 
> Hi Adrian,
> 
> The wolfSSL implementation does not support a TLS method range, such as 
> "TLSv1-", so that could be one of the causes. What seems strange is that 
> there is no warning message: "WARNING:tls_wolfssl:tls_get_method: wolfSSL 
> does not support method range specification" which should be thrown in such 
> cases.
> 
> Regards,
> 
> -- 
> Vlad Patrascu
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
> 
> On 12.08.2021 20:12, Adrian Georgescu wrote:
>> Hi,
>> 
>> I am using the latest 3.2.0 build with the old TLS configuration, with the 
>> aim to try out Wolf SSL stack.
>> 
>> But while the config check passed, the server does not start with the old 
>> configuration:
>> 
>> loadmodule “tls_mgm.so"
>> loadmodule “tls_wolfssl.so"
>> modparam("tls_mgm", "client_tls_domain_avp", "tls_client_domain")
>> modparam("tls_mgm", "tls_library", "auto”)
>> 
>> modparam("tls_mgm", "server_domain","ag-projects-server")
>> modparam("tls_mgm", "match_ip_address", "[ag-projects-server]*")
>> modparam("tls_mgm", "match_sip_domain", 
>> "[ag-projects-server]ag-projects.com")
>> modparam("tls_mgm", "tls_method",   "[ag-projects-server]TLSv1-")
>> modparam("tls_mgm", "certificate",  
>> "[ag-projects-server]/etc/opensips/tls/ag-projects.crt")
>> modparam("tls_mgm", "private_key",  
>> "[ag-projects-server]/etc/opensips/tls/ag-projects.key")
>> modparam("tls_mgm", "ca_list",  
>> "[ag-projects-server]/etc/opensips/tls/ca-list.pem")
>> modparam("tls_mgm", "ca_dir",   "[ag-projects-server]/etc/ssl/certs")
>> modparam("tls_mgm", "verify_cert",  "[ag-projects-server]1")
>> modparam("tls_mgm", "require_cert", "[ag-projects-server]0")
>> 
>> modparam("tls_mgm", "client_domain","ag-projects-client")
>> modparam("tls_mgm", "match_ip_address", "[ag-projects-client]*")
>> modparam("tls_mgm", "match_sip_domain", 
>> "[ag-projects-client]ag-projects.com")
>> modparam("tls_mgm", "tls_method",   "[ag-projects-client]TLSv1-")
>> modparam("tls_mgm", "certificate",  
>> "[ag-projects-client]/etc/opensips/tls/ag-projects.crt")
>> modparam("tls_mgm", "private_key",  
>> "[ag-projects-client]/etc/opensips/tls/ag-projects.key")
>> modparam("tls_mgm", "ca_list",  
>> "[ag-projects-client]/etc/opensips/tls/ca-list.pem")
>> modparam("tls_mgm", "ca_dir",   "[ag-projects-client]/etc/ssl/certs")
>> modparam("tls_mgm", "verify_cert",  "[ag-projects-client]1")
>> modparam("tls_mgm", "require_cert", "[ag-projects-client]0”)
>> 
>> 
>> Aug 12 18:51:14 live01 opensips[6455]: Aug 12 18:51:14 [6455] 
>> DBG:core:set_mod_param_regex: tls_mgm matches module tls_mgm
>> Aug 12 18:51:14 live01 opensips[6455]: Aug 12 18:51:14 [6455] 
>> DBG:core:set_mod_param_regex: found  in module tls_mgm 
>> [/usr/lib/x86_64-linux-gnu/opensips/modules/]
>> Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
>> DBG:core:solve_module_dependencies: solving dependency tls_mgm -> module 
>> tls_wolfssl
>> Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
>> DBG:core:solve_module_dependencies: solving dependency tls_mgm -> module 
>> tls_openssl
>> Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
>> DBG:core:solve_module_dependencies: module tls_mgm soft-depends on module 
>> tls_openssl, and it was not loaded -- continuing
>> Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
>> DBG:core:solve_module_dependencies: solving dependency proto_tls -> module 
>> tls_mgm
>> Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
>> DBG:core:init_mod: initializing module tls_mgm
>> Aug 12 18:51:15 live01 opensips[6455]: Aug 12 

Re: [OpenSIPS-Users] 3.2.0 TLS MGM module does not load 3.1.0 domain configuration

[Adrian Georgescu]

H Ovidiu,

I set it up explicitly now but I get the same result, I tried different domains 
or combination but any definition fails to load.

Aug 12 21:10:30 live01 /usr/sbin/opensips[10920]: 
ERROR:tls_mgm:init_tls_domains: Failed to init TLS domain 'msteams-server'
Aug 12 21:10:30 live01 /usr/sbin/opensips[10920]: ERROR:core:init_mod: failed 
to initialize module tls_mgm
Aug 12 21:10:30 live01 /usr/sbin/opensips[10920]: ERROR:core:main: error while 
initializing modules

loadmodule "tls_mgm.so"
modparam("tls_mgm", "tls_library", "wolfssl")

Regards,
Adrian

> On 12 Aug 2021, at 16:03, Ovidiu Sas  wrote:
> 
> Hello Adrian,
> 
> I managed to use wolfssl by forcing it:
>  modparam("tls_mgm", "tls_library", "wolfssl")
> 
> I haven't tested the auto mode ...
> 
> -ovidiu
> 
> On Thu, Aug 12, 2021 at 2:59 PM Adrian Georgescu  wrote:
>> 
>> After more digging I discovered that this behaviour does not happen when 
>> loading tls_openssl module.
>> 
>> tls_openssl loads fine this configuration but tls_wolfssl does not.
>> 
>>> On 12 Aug 2021, at 14:12, Adrian Georgescu  wrote:
>>> 
>>> Hi,
>>> 
>>> I am using the latest 3.2.0 build with the old TLS configuration, with the 
>>> aim to try out Wolf SSL stack.
>>> 
>>> But while the config check passed, the server does not start with the old 
>>> configuration:
>>> 
>>> loadmodule “tls_mgm.so"
>>> loadmodule “tls_wolfssl.so"
>>> modparam("tls_mgm", "client_tls_domain_avp", "tls_client_domain")
>>> modparam("tls_mgm", "tls_library", "auto”)
>>> 
>>> modparam("tls_mgm", "server_domain","ag-projects-server")
>>> modparam("tls_mgm", "match_ip_address", "[ag-projects-server]*")
>>> modparam("tls_mgm", "match_sip_domain", 
>>> "[ag-projects-server]ag-projects.com")
>>> modparam("tls_mgm", "tls_method",   "[ag-projects-server]TLSv1-")
>>> modparam("tls_mgm", "certificate",  
>>> "[ag-projects-server]/etc/opensips/tls/ag-projects.crt")
>>> modparam("tls_mgm", "private_key",  
>>> "[ag-projects-server]/etc/opensips/tls/ag-projects.key")
>>> modparam("tls_mgm", "ca_list",  
>>> "[ag-projects-server]/etc/opensips/tls/ca-list.pem")
>>> modparam("tls_mgm", "ca_dir",   
>>> "[ag-projects-server]/etc/ssl/certs")
>>> modparam("tls_mgm", "verify_cert",  "[ag-projects-server]1")
>>> modparam("tls_mgm", "require_cert", "[ag-projects-server]0")
>>> 
>>> modparam("tls_mgm", "client_domain","ag-projects-client")
>>> modparam("tls_mgm", "match_ip_address", "[ag-projects-client]*")
>>> modparam("tls_mgm", "match_sip_domain", 
>>> "[ag-projects-client]ag-projects.com")
>>> modparam("tls_mgm", "tls_method",   "[ag-projects-client]TLSv1-")
>>> modparam("tls_mgm", "certificate",  
>>> "[ag-projects-client]/etc/opensips/tls/ag-projects.crt")
>>> modparam("tls_mgm", "private_key",  
>>> "[ag-projects-client]/etc/opensips/tls/ag-projects.key")
>>> modparam("tls_mgm", "ca_list",  
>>> "[ag-projects-client]/etc/opensips/tls/ca-list.pem")
>>> modparam("tls_mgm", "ca_dir",   
>>> "[ag-projects-client]/etc/ssl/certs")
>>> modparam("tls_mgm", "verify_cert",  "[ag-projects-client]1")
>>> modparam("tls_mgm", "require_cert", "[ag-projects-client]0”)
>>> 
>>> 
>>> Aug 12 18:51:14 live01 opensips[6455]: Aug 12 18:51:14 [6455] 
>>> DBG:core:set_mod_param_regex: tls_mgm matches module tls_mgm
>>> Aug 12 18:51:14 live01 opensips[6455]: Aug 12 18:51:14 [6455] 
>>> DBG:core:set_mod_param_regex: found  in module tls_mgm 
>>> [/usr/lib/x86_64-linux-gnu/opensips/modules/]
>>> Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
>>> DBG:core:solve_module_dependencies: solving dependency tls_mgm -> module 
>>> tls_wolfssl

Re: [OpenSIPS-Users] auth_db module in 3.2.2

[Adrian Georgescu]

After removing the ha1b column, I am now getting the following errors and 
authentication does not work:

Aug 12 20:51:59 live01 /usr/sbin/opensips[10064]: 
ERROR:db_mysql:db_mysql_store_result: driver error: Commands out of sync; you 
can't run this command now
Aug 12 20:51:59 live01 /usr/sbin/opensips[10064]: ERROR:auth_db:get_ha1: failed 
to query database
Aug 12 20:52:00 live01 /usr/sbin/opensips[10057]: 
ERROR:db_mysql:db_mysql_store_result: driver error: Commands out of sync; you 
can't run this command now
Aug 12 20:52:00 live01 /usr/sbin/opensips[10057]: ERROR:auth_db:get_ha1: failed 
to query database

auth_db module configuration:

modparam("auth_db", "calculate_ha1", 0)
modparam("auth_db", "password_column",   "ha1")
modparam("auth_db", "user_column",   "username")
modparam("auth_db", "domain_column", "domain”)

What can be the reason for this?

Regards,
Adrian



> On 12 Aug 2021, at 13:04, Liviu Chircu  wrote:
> 
> On 12.08.2021 18:36, Adrian Georgescu wrote:
>> The auth_db module has some dramatic changes which are either undocumented 
>> or not backwards compatible and is unclear how to handle this.
>> 
>> https://opensips.org/docs/modules/3.1.x/auth_db.html#param_password_column_2 
>> <https://opensips.org/docs/modules/3.1.x/auth_db.html#param_password_column_2>Hi
>>  Adrian,
> 
> Indeed, with the addition of RFC 8760 support (support for SHA-256 and 
> SHA-512-256 auth algorithms), me and Maksym Sobolyev decided to try and 
> remove the "ha1b" feature, originally designed to accommodate some broken SIP 
> UAs who cannot follow the basic SIP authentication spec.  The feature had 
> been in there since the very beginnings, and we were not sure if anyone is 
> really benefiting from it anymore nowadays.
> 
> A strong reason for removing "ha1b" was the sheer number of hashes to be 
> stored per subscriber.  Since we now have 3 algorithms (MD5, SHA-256, 
> SHA-512-256), there are 3 hash-columns to store.  With the "ha1b" feature, 
> there would be 2 x 3 = 6 hashes in total to store, per user.  So you can see 
> where this is going: "Can we get away with dropping ha1b and storing half the 
> data per user?" ... was the big question.
> 
> Still, we agreed that if there is still enough traction for the "ha1b" 
> feature from the community, we can easily re-add the ha1b logic and 3 more 
> columns to the table and backport everything to 3.2.  It's a trivial task, 
> frankly.
> 
> The big question is: on your platform(s), can you control the software in all 
> SIP UAs that incorrectly include "realm" information in the "username" field 
> (which should really be just the user's name!) and fix the problem on the 
> phone side?
> 
> PS: I noticed the 3.2 migration page is missing any info on ha1b.  Will get 
> it fixed soon, depending on the outcome of the discussion.
> 
> Best Regards,
> 
> -- 
> Liviu Chircu
> www.twitter.com/liviuchircu <http://www.twitter.com/liviuchircu> | 
> www.opensips-solutions.com <http://www.opensips-solutions.com/>
> OpenSIPS Summit 2021 Distributed | www.opensips.org/events 
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Re: [OpenSIPS-Users] 3.2.0 TLS MGM module does not load 3.1.0 domain configuration

[Adrian Georgescu]

This line looks suspicious as I have not loaded or specified anywhere 
tls_openssl.

Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
DBG:core:solve_module_dependencies: module tls_mgm soft-depends on module 
tls_openssl, and it was not loaded — continuing

Adrian


> On 12 Aug 2021, at 14:12, Adrian Georgescu  wrote:
> 
> Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
> DBG:core:solve_module_dependencies: module tls_mgm soft-depends on module 
> tls_openssl, and it was not loaded -- continuing

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Re: [OpenSIPS-Users] 3.2.0 TLS MGM module does not load 3.1.0 domain configuration

[Adrian Georgescu]

After more digging I discovered that this behaviour does not happen when 
loading tls_openssl module. 

tls_openssl loads fine this configuration but tls_wolfssl does not.

> On 12 Aug 2021, at 14:12, Adrian Georgescu  wrote:
> 
> Hi,
> 
> I am using the latest 3.2.0 build with the old TLS configuration, with the 
> aim to try out Wolf SSL stack.
> 
> But while the config check passed, the server does not start with the old 
> configuration:
> 
> loadmodule “tls_mgm.so"
> loadmodule “tls_wolfssl.so"
> modparam("tls_mgm", "client_tls_domain_avp", "tls_client_domain")
> modparam("tls_mgm", "tls_library", "auto”)
> 
> modparam("tls_mgm", "server_domain","ag-projects-server")
> modparam("tls_mgm", "match_ip_address", "[ag-projects-server]*")
> modparam("tls_mgm", "match_sip_domain", "[ag-projects-server]ag-projects.com")
> modparam("tls_mgm", "tls_method",   "[ag-projects-server]TLSv1-")
> modparam("tls_mgm", "certificate",  
> "[ag-projects-server]/etc/opensips/tls/ag-projects.crt")
> modparam("tls_mgm", "private_key",  
> "[ag-projects-server]/etc/opensips/tls/ag-projects.key")
> modparam("tls_mgm", "ca_list",  
> "[ag-projects-server]/etc/opensips/tls/ca-list.pem")
> modparam("tls_mgm", "ca_dir",   "[ag-projects-server]/etc/ssl/certs")
> modparam("tls_mgm", "verify_cert",  "[ag-projects-server]1")
> modparam("tls_mgm", "require_cert", "[ag-projects-server]0")
> 
> modparam("tls_mgm", "client_domain","ag-projects-client")
> modparam("tls_mgm", "match_ip_address", "[ag-projects-client]*")
> modparam("tls_mgm", "match_sip_domain", "[ag-projects-client]ag-projects.com")
> modparam("tls_mgm", "tls_method",   "[ag-projects-client]TLSv1-")
> modparam("tls_mgm", "certificate",  
> "[ag-projects-client]/etc/opensips/tls/ag-projects.crt")
> modparam("tls_mgm", "private_key",  
> "[ag-projects-client]/etc/opensips/tls/ag-projects.key")
> modparam("tls_mgm", "ca_list",  
> "[ag-projects-client]/etc/opensips/tls/ca-list.pem")
> modparam("tls_mgm", "ca_dir",   "[ag-projects-client]/etc/ssl/certs")
> modparam("tls_mgm", "verify_cert",  "[ag-projects-client]1")
> modparam("tls_mgm", "require_cert", "[ag-projects-client]0”)
> 
> 
> Aug 12 18:51:14 live01 opensips[6455]: Aug 12 18:51:14 [6455] 
> DBG:core:set_mod_param_regex: tls_mgm matches module tls_mgm
> Aug 12 18:51:14 live01 opensips[6455]: Aug 12 18:51:14 [6455] 
> DBG:core:set_mod_param_regex: found  in module tls_mgm 
> [/usr/lib/x86_64-linux-gnu/opensips/modules/]
> Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
> DBG:core:solve_module_dependencies: solving dependency tls_mgm -> module 
> tls_wolfssl
> Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
> DBG:core:solve_module_dependencies: solving dependency tls_mgm -> module 
> tls_openssl
> Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
> DBG:core:solve_module_dependencies: module tls_mgm soft-depends on module 
> tls_openssl, and it was not loaded -- continuing
> Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
> DBG:core:solve_module_dependencies: solving dependency proto_tls -> module 
> tls_mgm
> Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
> DBG:core:init_mod: initializing module tls_mgm
> Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
> INFO:tls_mgm:mod_init: initializing TLS management
> Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
> DBG:tls_mgm:load_info: 0 rows found in tls_mgm
> Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
> DBG:tls_mgm:load_info: 0 records found in tls_mgm
> Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
> INFO:tls_mgm:init_tls_dom: Processing TLS domain 'ag-projects-server'
> Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
> DBG:tls_mgm:init_tls_dom: no DH params file for tls domain 
> 'ag-projects-server' defined, using default '(null)'
> Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
> DBG:tls_mgm:init_tls_dom: cipher list null ... setting default
> Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
> NOTICE:tls_mgm:init_tls_dom: no crl for tls, using none
> Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
> ERROR:tls_mgm:init_tls_domains: Failed to init TLS domain 'ag-projects-server'
> Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
> ERROR:core:init_mod: failed to initialize module tls_mgm
> 
> Any ideas what am I doing wrong?
> 
> Adrian
> 
> 


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[OpenSIPS-Users] 3.2.0 TLS MGM module does not load 3.1.0 domain configuration

[Adrian Georgescu]

Hi,

I am using the latest 3.2.0 build with the old TLS configuration, with the aim 
to try out Wolf SSL stack.

But while the config check passed, the server does not start with the old 
configuration:

loadmodule “tls_mgm.so"
loadmodule “tls_wolfssl.so"
modparam("tls_mgm", "client_tls_domain_avp", "tls_client_domain")
modparam("tls_mgm", "tls_library", "auto”)

modparam("tls_mgm", "server_domain","ag-projects-server")
modparam("tls_mgm", "match_ip_address", "[ag-projects-server]*")
modparam("tls_mgm", "match_sip_domain", "[ag-projects-server]ag-projects.com")
modparam("tls_mgm", "tls_method",   "[ag-projects-server]TLSv1-")
modparam("tls_mgm", "certificate",  
"[ag-projects-server]/etc/opensips/tls/ag-projects.crt")
modparam("tls_mgm", "private_key",  
"[ag-projects-server]/etc/opensips/tls/ag-projects.key")
modparam("tls_mgm", "ca_list",  
"[ag-projects-server]/etc/opensips/tls/ca-list.pem")
modparam("tls_mgm", "ca_dir",   "[ag-projects-server]/etc/ssl/certs")
modparam("tls_mgm", "verify_cert",  "[ag-projects-server]1")
modparam("tls_mgm", "require_cert", "[ag-projects-server]0")

modparam("tls_mgm", "client_domain","ag-projects-client")
modparam("tls_mgm", "match_ip_address", "[ag-projects-client]*")
modparam("tls_mgm", "match_sip_domain", "[ag-projects-client]ag-projects.com")
modparam("tls_mgm", "tls_method",   "[ag-projects-client]TLSv1-")
modparam("tls_mgm", "certificate",  
"[ag-projects-client]/etc/opensips/tls/ag-projects.crt")
modparam("tls_mgm", "private_key",  
"[ag-projects-client]/etc/opensips/tls/ag-projects.key")
modparam("tls_mgm", "ca_list",  
"[ag-projects-client]/etc/opensips/tls/ca-list.pem")
modparam("tls_mgm", "ca_dir",   "[ag-projects-client]/etc/ssl/certs")
modparam("tls_mgm", "verify_cert",  "[ag-projects-client]1")
modparam("tls_mgm", "require_cert", "[ag-projects-client]0”)


Aug 12 18:51:14 live01 opensips[6455]: Aug 12 18:51:14 [6455] 
DBG:core:set_mod_param_regex: tls_mgm matches module tls_mgm
Aug 12 18:51:14 live01 opensips[6455]: Aug 12 18:51:14 [6455] 
DBG:core:set_mod_param_regex: found  in module tls_mgm 
[/usr/lib/x86_64-linux-gnu/opensips/modules/]
Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
DBG:core:solve_module_dependencies: solving dependency tls_mgm -> module 
tls_wolfssl
Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
DBG:core:solve_module_dependencies: solving dependency tls_mgm -> module 
tls_openssl
Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
DBG:core:solve_module_dependencies: module tls_mgm soft-depends on module 
tls_openssl, and it was not loaded -- continuing
Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
DBG:core:solve_module_dependencies: solving dependency proto_tls -> module 
tls_mgm
Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
DBG:core:init_mod: initializing module tls_mgm
Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
INFO:tls_mgm:mod_init: initializing TLS management
Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
DBG:tls_mgm:load_info: 0 rows found in tls_mgm
Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
DBG:tls_mgm:load_info: 0 records found in tls_mgm
Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
INFO:tls_mgm:init_tls_dom: Processing TLS domain 'ag-projects-server'
Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
DBG:tls_mgm:init_tls_dom: no DH params file for tls domain 'ag-projects-server' 
defined, using default '(null)'
Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
DBG:tls_mgm:init_tls_dom: cipher list null ... setting default
Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
NOTICE:tls_mgm:init_tls_dom: no crl for tls, using none
Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
ERROR:tls_mgm:init_tls_domains: Failed to init TLS domain 'ag-projects-server'
Aug 12 18:51:15 live01 opensips[6455]: Aug 12 18:51:15 [6455] 
ERROR:core:init_mod: failed to initialize module tls_mgm

Any ideas what am I doing wrong?

Adrian



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[OpenSIPS-Users] auth_db module in 3.2.2

[Adrian Georgescu]

The auth_db module has some dramatic changes which are either undocumented or 
not backwards compatible and is unclear how to handle this.

https://opensips.org/docs/modules/3.1.x/auth_db.html#param_password_column_2 


Aug 12 17:34:10 [3179] CRITICAL:core:yyerror: parse error in 
/etc/opensips/opensips.cfg.tmp:1170:20-21: Parameter  not 
found in module  - can't set
Aug 12 17:34:10 [3179] modparam("auth_db", "calculate_ha1", 0)
Aug 12 17:34:10 [3179] modparam("auth_db", "password_column",   "ha1")
Aug 12 17:34:10 [3179] modparam("auth_db", "password_column_2", "ha1b")
Aug 12 17:34:10 [3179] ^~
Aug 12 17:34:10 [3179] modparam("auth_db", "user_column",   "username")
Aug 12 17:34:10 [3179] modparam("auth_db", "domain_column", "domain")
Aug 12 17:34:10 [3179] ERROR:core:parse_opensips_cfg: bad config file (1 errors)

password_column_2 parameter has vanished in 3.2.2 and we relied heavily on its 
presence as it contained a recalculate hash including the domain name.

How should we deal with this?

Regards,
Adrian

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[OpenSIPS-Users] Usage of TLS require_cert and verify_cert settings

[Adrian Georgescu]

Hello,

I have a question.

I have the following TLS scenarios:

1) A local user for a domain I own, connects to my server using TLS. If the 
domain is local, I will authenticate the user against my database and I do not 
care if the user has a certificate
2) A remote server, connects to my server using TLS and pretends that is 
domainX.com. In such case, the only way to verify that this is true is by 
requiring a certificate and verify it

So there is a logic split between when to require and how to verify a 
certificate depending on the fact that we deal with a local user or a foreign 
domain.

I would like to know if is possible to set require_cert and verify_cert on the 
fly, while routing packets, instead of configuring them statically per domain.

Regards,
Adrian




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Re: [OpenSIPS-Users] OpenSIPS as simple SIP proxy

[Adrian Georgescu]

You can install a pre-configured SIP Proxy as a start point, like OpenSIPS.

Installation instructions are available.

here 

OpenSIPS configuration debian package is available here 
.

The package name is opensips-config-light

Regards,
Adrian


> On 13 Jan 2021, at 04:16, Conor Power  wrote:
> 
> Hi,
>Apologies for the noob question but I'm hoping someone can point me in the 
> right direction.
> 
> I am trying to use OpenSIPS as a simple proxy to proxy all calls to another 
> SIP endpoint and back again to the original client. The only role of the 
> OpenSIPS server is to function as the proxy and it is for all requests.
> 
> I have OpenSIPS up and running and can see the requests coming inbound using 
> ngrep but I've had no success proxying the requests.
> 
> I added a sethostport() call in the config file but really am not sure where 
> or how it fits in.
> 
> If someone might point me to a simple config file that would be used for such 
> a proxy setup, it would be greatly appreciated.
> 
> Conor
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Re: [OpenSIPS-Users] OpenSIPS 3.1 & NAT issues

[Adrian Georgescu]

Google search for SIP ALG problem to see if this is relevant for your case.

Regards,
Adrian


> On 13 Jan 2021, at 13:08, Mark Allen  wrote:
> 
> Hi all - I've been banging my head against this but not succeeding.
> 
> Our setup...
> 
> UAC   192.168.x.x
>   | 
> Router5.x.x.x
>   |
> (internet)
>   | 
> Firewall  46.x.x.x maps
>   |   directly to
> OpenSIPS  192.168.x.x  Mid-registrar
>   |
> Asterisk  192.168.x.x
> 
> 
> Current situation: 
> - UAC can register on Asterisk via OpenSIPS
> - UAC can call destination registered on Asterisk on local n/w to Asterisk box
> - Destination extension rings and can pick up call
> - There is no audio either way & call drops after about 30 secs (Asterisk 
> kills call with "Requested channel not available" because not RTP traffic is 
> reaching destination)
> 
> I have tried passing audio through Mediaproxy on OpenSIPS box but with no 
> success. Using Wireshark I can see RTP traffic initiated at both ends, but it 
> doesn't reach the other end either way.
> 
> Is there some definitive guide to setting this up correctly or are there 
> specific steps that I need to follow? 
> 
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Re: [OpenSIPS-Users] Mediaproxy Relay start error - 'MediaRelayBase' is not defined

[Adrian Georgescu]

This was a bug.

You must update to the latest mediaproxy version:

sudo apt update 
sudo apt install mediaproxy-relay mediaproxy-common mediaproxy-dispatcher

Regards,
Adrian

> On 6 Jan 2021, at 12:59, Mark Allen  wrote:
> 
> Hi all - not sure what I'm missing here... 
> 
> I'm installing Mediaproxy onto our Debian Buster box which is also running 
> OpenSIPS 3.1 but when I start the relay I'm getting an error in syslog...
> 
> 15:40:07 opensipsx media-relay[4983]: INFO Starting MediaProxy Relay 4.0.4
> 15:40:07 opensipsx media-relay[4983]: INFO Set resource limit for maximum 
> open file descriptors to 11000
> 15:40:07 opensipsx media-relay[4983]: CRITICAL Failed to create MediaProxy 
> Relay: name 'MediaRelayBase' is not defined
> 15:40:07 opensipsx media-relay[4983]: ERRORTraceback (most recent call 
> last):#012ERROR  File "/usr/bin/media-relay", line 100, in 
> #012ERRORfrom mediaproxy.relay import MediaRelay#012ERROR 
>  File "/usr/lib/python3/dist-packages/mediaproxy/relay.py", line 290, in 
> #012ERRORclass MediaRelay(MediaRelayBase):#012ERROR
> NameError: name 'MediaRelayBase' is not defined
> 15:40:07 opensipsx systemd[1]: mediaproxy-relay.service: Main process exited, 
> code=exited, status=1/FAILURE
> 15:40:07 opensipsx systemd[1]: mediaproxy-relay.service: Failed with result 
> 'exit-code'.
> 
> I'm starting the relay with the command...
> 
> systemctl start mediaproxy-relay
> 
> ...I installed Mediaproxy using the Debian package using the instructions at 
> http://mediaproxy.ag-projects.com/installation-guide/ 
>  and 
> https://github.com/AGProjects/mediaproxy 
> . mediaproxy-dispatcher is starting 
> successfully.
> 
> In the /etc/mediaproxy/config.ini file - everything is left at the default 
> setting except for...
> 
> dispatchers = xxx.xxx.xxx.xxx
> advertised_ip = xxx.xxx.xxx.xxx
> 
> Certificates are in place in /etc/mediaproxy/tls
> 
> 
> Anybody got any ideas about where I've gone wrong
> 
> 
> 
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[OpenSIPS-Users] New MediaProxy release 4.0.4

[Adrian Georgescu]

Hello,

There is a new MediaProxy release compatible with Python 3.

The binary .deb package is available for Debian Sid in AG Projects Debian 
repository.

To install or upgrade see instructions from:

http://mediaproxy.ag-projects.com

mediaproxy (4.0.4) unstable; urgency=medium

  * Migration to Python 3
  * Fixed checking relay passport
  * Fix for fetching statistics 
  * Added gracefull shutdown option to systemd unit using reload command
  * Remove invalid argument to loseConnection
  * Fix removing expired sessions
  * Add SIP call-id to logging
  * Improve logging at start
  * Added libiptc-dev build dependency
  * Log which dispatchers will be used
  * Use local logger for traffic
  * Log call statistics
  * Fixed logging of radius configurations
  * Improve log line for used TLS certificate 
  * Allow use of configured dispatchers in Thor mode
  * Added logs about TLS certificates loaded at start
  * Log remote certificate common name for all TLS connections
  * Fixed sending back expired event to dispatcher

Regards,
Adrian


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Re: [OpenSIPS-Users] Sylk Mobile

[Adrian Georgescu]

Hi Maxim,

I am travelling between two continents on Saturday,

If everything goes well, I would like to be your guest in one of your next 
chronicles!

Adrian

> On 11 Aug 2020, at 15:38, Maxim Sobolev  wrote:
> 
> Interesting work Adrian! Any chance you can be interested in coming over to 
> our SIP Chronicles videocast to talk about it and perhaps do a live demo? 😀
> 
> https://www.youtube.com/playlist?list=PL-U7hOT8zFXoSMgHLfVj_CX4MvFjD2gcj 
> <https://www.youtube.com/playlist?list=PL-U7hOT8zFXoSMgHLfVj_CX4MvFjD2gcj>
> 
> Let me know, we don't have anyone booked yet for this coming Saturday.
> 
> Thanks!
> 
> -Max
> 
> -Max
> 
> On Thu., Aug. 6, 2020, 11:07 p.m. Adrian Georgescu,  <mailto:a...@ag-projects.com>> wrote:
> Hi,
> 
> We just published Sylk Mobile, a react-native mobile client for Android and 
> iOS.
> 
> The server side is running OpenSIPS + Janus.
> 
> https://lists.ag-projects.com/pipermail/sipbeyondvoip/2020-August/003469.html 
> <https://lists.ag-projects.com/pipermail/sipbeyondvoip/2020-August/003469.html>
> 
> Enjoy!
> Adrian
> 
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[OpenSIPS-Users] Sylk Mobile

[Adrian Georgescu]

Hi,

We just published Sylk Mobile, a react-native mobile client for Android and iOS.

The server side is running OpenSIPS + Janus.

https://lists.ag-projects.com/pipermail/sipbeyondvoip/2020-August/003469.html 


Enjoy!
Adrian

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Re: [OpenSIPS-Users] Push notifications server

[Adrian Georgescu]

Download instructions are listed here:

https://lists.ag-projects.com/pipermail/sipbeyondvoip/2020-June/003458.html 
<https://lists.ag-projects.com/pipermail/sipbeyondvoip/2020-June/003458.html>


> On 20 Jun 2020, at 12:50, Johan De Clercq  wrote:
> 
> Adrian, 
> Where can I download this ?
> 
> Outlook voor iOS <https://aka.ms/o0ukef> downloaden
> Van: Users  namens Adrian Georgescu 
> 
> Verzonden: woensdag, juni 17, 2020 5:53 PM
> Aan: OpenSIPS users mailling list
> Onderwerp: [OpenSIPS-Users] Push notifications server
>  
> Hello,
> 
> We just made public a mobile push notification server that may help in 
> various scenarios.
> 
> An integration guide for OpenSIPS is available in the source code.
> 
> https://ag-projects.com/news/sylk-pushserver/ 
> <https://ag-projects.com/news/sylk-pushserver/>
> 
> Regards,
> Adrian
> 
> 
>  
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[OpenSIPS-Users] Push notifications server

[Adrian Georgescu]

Hello,

We just made public a mobile push notification server that may help in various 
scenarios.

An integration guide for OpenSIPS is available in the source code.

https://ag-projects.com/news/sylk-pushserver/

Regards,
Adrian


 
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Re: [OpenSIPS-Users] Join us for SIP Chronicles Live #1, featuring Giovanni Maruzzelli

[Adrian Georgescu]

You can try out SylkServer / sip2sip.info 

Adrian

> On 21 Apr 2020, at 17:48, Maxim Sobolev  wrote:
> 
> Thanks Bogdan, I am glad that you liked the idea! Yes, very good question. We 
> will have a slot (or few) where questions from the audience can be answered 
> interactively. Originally we were planning to take questions over YouTube 
> chat, but maybe it would be also cool if Giovanni can deploy his cool phone 
> so people can actually call in and ask? At which point we could also publish 
> a SIP URI for anyone to  ring in directly as well and drill speaker on his 
> answers.
> 
> Eventually I hope to feel brave enough to deploy Jitsi Meet, but probably not 
> until this whole ordeal is over unless I get some more help from a community, 
> which is also an option. :)
> 
> -Max
> 
> On Tue, Apr 21, 2020 at 9:00 AM Bogdan-Andrei Iancu  > wrote:
> Hi Maxim,
> 
> Great idea, let's keep the communities connected and up to date - after all 
> this is what we do - we do communication systems :)
> 
> Giovanni, I will be there !
> 
> BTW, is this an interactive session, in the way that questions can be asked?
> 
> Best regards,
> Bogdan-Andrei Iancu
> 
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com 
> 
> On 4/20/20 9:59 PM, Maxim Sobolev wrote:
>> Dear Real-Time Friends and Colleagues!
>> 
>> As many of you we have been totally devastated that we will have no chance 
>> to see you in the next few months to come. :-/ Some people in the community 
>> believe it might be years. I don’t necessarily agree with that opinion 
>> myself.
>> 
>> Over the course of the last few years our team had a great time extending 
>> live coverage for some of those events that have been affected, got some 
>> experience and equipment. Instead of just waiting for the virus to clear, we 
>> decided to organize a series of bi-weekly live casts with some of the 
>> speakers that we have hoped to see at those events presenting their latest 
>> developments live and then answering questions from the audience. 
>> 
>> So without further ado, let me introduce our first guest Giovanni 
>> Maruzzelli, who is going to introduce his newest project SaraPhone 
>> (https://github.com/gmaruzz/saraphone 
>> ). 
>> 
>> Join us this Saturday, April 25th 4:30pm UTC and get a chance to ask 
>> Giovanni a question about his project live:
>> 
>> https://youtu.be/mF9elIcVGE8 
>> 
>> Or if you miss that opportunity, you can always watch the recording later on 
>> Sippy Labs channel on YouTube and email Giovanni your question at 
>> mailto:gmar...@gmail.com>>.
>> 
>> SaraPhone is a bare bone SIP WebRTC phone, complete with most features real 
>> companies want to use in real world: HotDesking, Redial, BLFs, MWI, DND, 
>> PhoneBook, Hold, Transfer, Mute, Attended Transfer, Notifications, running 
>> on all Browsers both on Desktop and SmartPhone.
>> 
>> SaraPhone is fully integrated with FusionPBX, the full-featured domain based 
>> multi-tenant PBX and voice switch for FreeSwitch.
>> 
>> Based on SIP.js, SaraPhone works with all WebRTC compliant SIP proxies, 
>> gateways, and servers (Asterisk, OpenSIPS, Kamailio, Janus, etc).
>> 
>> Initial author is Giovanni Maruzzelli, and SaraPhone gets its name from 
>> Giovanni's wife, Sara Hosseini.
>> 
>> In addition to providing all of the usual DeskPhone functionality, SaraPhone 
>> got:
>> 
>> Desktop Notification for Incoming Calls
>> Live MWI update
>> Real Time BLFs status update
>> BLF click to call
>> Caller Name and Number Display
>> Call Error Cause Display
>> AutoAnswer
>> Network Disconnect Reload
>> Show and Set Caller-ID (incoming-outbound)
>> 
>> Stay healthy, optimistic and productive! Also share, like and subscribe. See 
>> you soon!!!
>> 
>> Regards,
>> 
>> Max
>> 
>> 
>> ___
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>> Users@lists.opensips.org 
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users 
>> 
> 
> 
> 
> -- 
> Maksym Sobolyev
> Sippy Software, Inc.
> Internet Telephony (VoIP) Experts
> Tel (Canada): +1-778-783-0474
> Tel (Toll-Free): +1-855-747-7779
> Fax: +1-866-857-6942
> Web: http://www.sippysoft.com 
> MSN: sa...@sippysoft.com 
> Skype: SippySoft
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[OpenSIPS-Users] New MediaProxy release 3.0.0

[Adrian Georgescu]

There is new MediaProxy release compatible with OpenSIPS 3.1.

Binary packages are available for Debian (Sid and Buster) and Ubuntu (Eoan and 
Bionic) 

For how to update the software see:

http://mediaproxy.ag-projects.com/download/

Changelog

mediaproxy 3.0.0
.
 * Renewed TLS certificates for another 10 years
 * Avoid leaking variable outside list comprehension
 * Updated opensips interface to use json
 * Renamed configuration_filename to configuration_file
 * Renamed system_config_directory to config_directory
 * Streamlined error logging for the OpenSIPS MI interface
 * Use new runtime directory
 * Adapted to logging API changes in python-application 2.8.0
 * Adapted to process API changes in python-application 2.8.0
 * Adjusted logging prefixes
 * Refactored the media-dispatcher and media-relay scripts
   - Use the argparse module for parsing command line arguments
   - Replaced --pid with --config-dir and --runtime-dir options
   - Added --debug and --debug-memory options
   - Added --systemd command line option
 * Updated manual pages
 * Added code to wait for the network to be available when starting
 * Adjusted runtime directory in sample config file
 * Update license files
 * Refactored setup.py
 * Refactored the debian build system
   - Increased debian compatibility level to 11
   - Removed no longer necessary files
   - Replaced the mediaproxy-common-dbg package with a dbgsym package
   - Removed no longer needed version checks from some dependencies
   - Switched to using the pybuild build system
   - Simplified the debian rules file
 * Switched from init scripts to systemd service files
 * Fixed pidfile argument to process.daemonize()
 * Align method signature with parent
 * Refactored code to simplify future changes to logging
 * Fixed message when connection is closed cleanly
 * Changed management interface name
 * Removed redundant comment
 * Align method signature with parent class
 * Removed unnecessary try/finally
 * Set attribute in __init__
 * Use super to call parent to avoid dumb IDE warning
 * Simplified conditional import
 * Use the protocol's delimiter instead of hardcoded values
 * Fixed dialog ids in tests
 * Removed redundant code
 * Changed ports used by tests to be different from those used by mediaproxy
 * Updated boring file
 * Improved logging
 * Fixed attribute name
 * Fixed some PEP-8 violations
 * Removed no longer needed future import
 * Removed unnecessary docstrings from top level scripts
 * Removed unnecessary variable
 * Added code to toggle verbose logging by sending SIGUSR1 to the process
 * Simplified signal handlers
 * Fixed issue with pickling RelaySession
 * Fixed exception handling
 * Fixed graceful shutdown when running with SIPThor
 * Made MANIFEST.in more explicit and avoid unnecessary prune commands
 * Pass command line arguments from build_inplace to setup.py
 * Cleanup after build_inplace
 * Removed commented out variable in debian rules
 * Explicitly use python2 in shebang lines
 * Updated tinyca URL
 * Updated copyright years
 * Increased debian standards version to 4.5.0


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Re: [OpenSIPS-Users] Remote Control Advice

[Adrian Georgescu]

You need an end-point device to initiate the call. The Proxy is just in the 
middle of the call and not the right place to start a call.

Adrian 

> On 10 Dec 2019, at 13:17, Mark Farmer  wrote:
> 
> Hi everyone
> 
> Let's say that I have an OpenSIPS box with phones registered and a home grown 
> web interface via which I need to initiate a call between 2 physical phones. 
> What I'm looking for is a way for the web interface to ask OpenSIPS to 
> initiate that call remotely.
> 
> The web GUI is (in this case) NOT an endpoint (webrtc etc) - just a mechanism 
> to initiate the call.
> 
> How can I go about doing this? Is it a case of using a SIP library to send a 
> SIP invite to OpenSIPS with the correct header info or is there another 
> mechanism that I can use?
> 
> Many thanks
> Mark.
> 
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Re: [OpenSIPS-Users] How to limit parallel calls duration of prepaid customers?

[Adrian Georgescu]

Call control does handle the ’special’ case of parallel calls for a  user. When 
balance gets to zero all outgoing calls are being cut add the same time. There 
is nothing special the module work with N parallel calls, not just 1.

Adrian

> On 16 Nov 2019, at 13:47, Dipteshkumar Patel  
> wrote:
> 
> Hello Team,
> 
> I want to use opensips as a pbx system. I have prepaid customers so how can i 
> manage prepaid calls scheduling based on customers' balance(dialog timeout). 
> As specially in case we have parallel calls of a user. I found call-control 
> from ag-projects for that. 
> 
> Can you please suggest any other possible way to implement this feature?
> 
> Thanks & Regards
> Diptesh Patel
> Software Developer 
> Ecosmob Technologies Ltd, 
> Ahmedabad
> Mo:+919898962659
> 
> Disclaimer
> In addition to generic Disclaimer which you have agreed on our website, any 
> views or opinions presented in this email are solely those of the originator 
> and do not necessarily represent those of the Company or its sister concerns. 
> Any liability (in negligence, contract or otherwise) arising from any third 
> party taking any action, or refraining from taking any action on the basis of 
> any of the information contained in this email is hereby excluded.
> 
> Confidentiality
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> of the addressee(s) and contains information that is PRIVILEGED AND 
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Re: [OpenSIPS-Users] Im with message.

[Adrian Georgescu]

SIP MESSAGE has nothing to do with SIP SIMPLE suite of standards.

SIP SIMPLE is a suite of standards for session based IM and file transfers 
using MSRP media plane and Presence. Presence requires contacts management for 
which XCAP server is used.  XCAP is built on top of an HTTP server by serving 
XML documents defined for contacts and policies used by Presence. These 
documents can be changed by a SIP client with XCAP support.

More information about the SIP SIMPLE standards are found here:

https://datatracker.ietf.org/wg/simple/documents/ 


Adrian


> On 23 Nov 2018, at 11:35, Johan De Clercq  wrote:
> 
> Okay, so I just can handle it trhough my normal script as I would do with an 
> initial INVITE ?
> Secondly, when exactly is XCAP needed ?
> And where does this fit with SIP SIMPLE ?
> 
> I am really sorry but I am not really used to presence and IM.
> 
> Op vr 23 nov. 2018 om 14:45 schreef Bogdan-Andrei Iancu  >:
> No :)
> Bogdan-Andrei Iancu
> 
> OpenSIPS Founder and Developer
>   http://www.opensips-solutions.com 
> OpenSIPS Bootcamp 2018
>   http://opensips.org/training/OpenSIPS_Bootcamp_2018/ 
> 
> On 11/23/2018 08:41 AM, Johan De Clercq wrote:
>> Hi, I need to be able to handle SIP MESSAGE for IM. Does this mean that I 
>> need to setup openxcap and use xcap client module in my script?
>> 
>> Best regards,
>> 
>> Outlook voor iOS  downloaden
>> 
>> 
>> ___
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users 
>> 
> 
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Re: [OpenSIPS-Users] Im with message.

[Adrian Georgescu]

SIP MESSAGE uses only the SIP proxy, you don’t need other components.

Adrian

> On 23 Nov 2018, at 03:41, Johan De Clercq  wrote:
> 
> Hi, I need to be able to handle SIP MESSAGE for IM. Does this mean that I 
> need to setup openxcap and use xcap client module in my script?
> 
> Best regards,
> 
> Outlook voor iOS  downloaden
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Re: [OpenSIPS-Users] Astricon: OpenSource Soiree

[Adrian Georgescu]

Great, see you there!

Adrian

> On 4 Oct 2018, at 11:54, Giovanni Maruzzelli  wrote:
> 
> Hello friends and collegues,
> 
> A nice get together has been organized for evening Wednesday 10th at 7pm
> 
> The idea is to have drinks and chitchat about opensource telephony and 
> webrtc, and all opensource project founders, members, and practitioners are 
> invited.
> 
> Please join me, as distinguished participants too!
> 
> Check it out at  http://party.officering.com 
> 
> (no animals has been harmed in this mail)
> 
> -giovanni
> 
> --
> 
> Sincerely,
> 
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
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[OpenSIPS-Users] New CDRTool release 9.6

[Adrian Georgescu]

There is a new release of CDRTool that addresses interoperability with latest 
PHP versions.

Changelog

   * Updated syntax and formatting
   * Removed short opening tags
   * Updated smarty dependency
   * Fixed update query causing errors in mysql strict
   * Fixed login when yubikey is not used
.
   [ Web Interface ]
.
   * Fixed conflicting 'arrow' classes
   * Added support for GTID in replication overview

http://cdrtool.ag-projects.com

Regards,
Adrian



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Re: [OpenSIPS-Users] OpenXCAP on Debian...

[Adrian Georgescu]

Can you paste the full output of your commands?

Regards,
Adrian


> On 6 Dec 2017, at 14:11, Boris via Users  wrote:
> 
> I just tried to install OpenXCAP on Debian Stretch and was having issues 
> because of missing dependencies (python-twisted-web2). I then tried on both 
> Jessie and Wheezy, but each of those were exposing different dependency 
> problems.
> 
> Was anybody successful recently in installing OpenXCAP on Debian? Not from 
> source but from binary packages?
> 
> Thanks
> Boris
> 
> 
> 
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Re: [OpenSIPS-Users] [SR-Users] ClueCon Slides: Industrial Grade FreeSWITCH, Scaling, Balancing, High Availability, SIP, WebRTC

[Adrian Georgescu]

Giovanni,

The creature on page 55 and 56 is a dude, man!

Regards,
Adrian

> On 11 Aug 2017, at 12:08, jungle Boogie  wrote:
> 
> On 10 August 2017 at 11:17, Giovanni Maruzzelli  wrote:
>> Industrial Grade FreeSWITCH Scaling, Balancing and High Availability for SIP
>> and WebRTC
>> 
>> Scaling your FreeSWITCH platform to serve a growing user base is a critical
>> challenge. We'll go through the best techniques, practices, and
>> implementations for Voice and Video Calls, Conferencing, WebRTC, SIP,
>> Chatting, Presence and Instant Messaging
> 
> 
> Any chance we can make a "papers" section in confluence with a link to
> it from freeswitch.org
> 
> OpenBSD does this with their presentations:
> https://www.openbsd.org/events.html
> 
> Nice way to list events and it can be referenced in emails, the
> conference call, etc as extra reading material.
> 
>> 
>> Slides can be downloaded from http://opentelecom.it/cluecon2017.pdf
>> 
>> Enjoy ClueCon!
>> 
>> -giovanni
>> 
> 
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[OpenSIPS-Users] New MediaProxy Release version 2.6.6

[Adrian Georgescu]

Hello,

There is a new Mediaproxy software available

Changelog mediaproxy (2.6.6)

   * Updated list of non-routable IP addresses and the corresponding RFC

The software is packaged for latest Debian and Ubuntu versions.

To update or install the software follow the instructions at

http://mediaproxy.ag-projects.com 

Regards,
Adrian



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[OpenSIPS-Users] new CDRTool release 9.5.0

[Adrian Georgescu]

Hello,

The software includes bug fixes, better SIP tracing and compatibility with 
latest PHP version(s) >= 5.4.

Debian packages are available for Debian Jessie, Wheezy and Unstable.

The software can be downloaded from:

http://cdrtool.ag-projects.com

Changelog

cdrtool (9.5.0) unstable; urgency=medium

  * Added mysqli database backend
  * Reworked Debian packaging
  * Switched to Net/URL2
  * Update PEAR to use new syntax (prevents warnings)
  * Fixed most PHP warning messages
  * Cleaned code
  * Formatted code
  * Added systemd service

  [ Web Interface ]

  * Fixed search in logging screen
  * Fixed updating subscriber ACLs
  * Updated trace links in sip settings to use a new window for each trace
  * Updated title in sip settings page
  * Updated DNS error messages
  * Added SQL debug printer
  * Updated SIP Trace to look better

Regards,
Adrian


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Re: [OpenSIPS-Users] Mediaproxy hanging sessions on high load

[Adrian Georgescu]

Perhaps your virtual machine simply cannot handle the load. The commands to 
close sessions may also be dropped or lost under such environment.

Adrian


> On 16 Mar 2017, at 11:22, Daniel Zanutti  wrote:
> 
> Hi Dan
> 
> Looks like this problem is only happening on virtual machines, not on 
> physical machines. And only while they are on high load.
> 
> But i'm not sure about the kernel rule, is there any way to check it?
> 
> Please take a look at this case, this Relay will never halt because there are 
> more than 3k sessions that will never finish internally (the call has already 
> hangup hours ago):
> 
> 8 2.2.2.2 2.6.1   44h01'05"
> 112.03kbps3045
> audio 3045Halting
> 
> Some of these calls:
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 728   From: 22@4.4.4.4
> To: 3...@sip.aaa.com.br 
>   6.6.6.6:55632    2.2.2.2:46640 
>    2.2.2.2:46866    
> 7.7.7.7:4170  active  G729audio   21h35'34" 
>   0   0
> 729   From: 222@4.4.4.4:5064
> To: 3...@sip.aaa.com.br 
>   6.6.6.6:34908   2.2.2.2:58158   2.2.2.2:54372   7.7.7.7:16846   
> active  G729audio   16h11'51"   0   0
> 730   From: @4.4.4.4
> To: 3...@sip.aaa.com.br 
>   6.6.6.6:46324    2.2.2.2:50156 
>    2.2.2.2:48182    
> 7.7.7.7:18516    active  G729audio   19h45'38" 
>   0   0
> 731   From: 22@4.4.4.4:5061
> To: ...@sip.aaa.com.br 
>   6.6.6.6:54800   2.2.2.2:43998   2.2.2.2:46144   7.7.7.7:12360   
> active  G729audio   19h09'41"   0   0
> 732   From: 222@4.4.4.4 
> To: 3...@sip.aaa.com.br 
>   6.6.6.6:18854    2.2.2.2:51924 
>    2.2.2.2:40512    
> 7.7.7.7:4200  active  G729audio   19h37'59" 
>   0   0
> 
> Is there any way to drop these sessions? Maybe using the internal timeout 
> system of mediaproxy?
> 
> If you could take a look personally, we could negotiate an hourly rate.
> 
> Thanks again
> 
> 
> 
> On Thu, Mar 16, 2017 at 10:54 AM, Dan Pascu  > wrote:
> 
> One thing came to mind. A case when the relay could get overloaded is if a 
> lot of clients start sessions and only one endpoint sends media. That is the 
> only case where the relay would have to deal with the media traffic itself 
> and having hundreds of such sessions at the same time could overload the 
> relay.
> 
> The way the relay works is for each call it starts listening on 4 ports (2 
> for RTP and 2 for RTCP). Each endpoint will send 2 streams (1 RTP one RTCP) 
> and initially the relay will just listen on these ports and when it receives 
> data it learns the endpoint's address. After it learns both endpoint's 
> addresses, it adds a conntrack rule in the kernel to allow the kernel to 
> directly relay the media streams between the endpoints and it will never see 
> a media packet from the endpoints again until the call ends. This allows for 
> very efficient data forwarding because it's done entirely in the kernel with 
> no data being transferred from kernel to user-space and back like traditional 
> solutions. We have seen media relays handling hundreds of calls at a time 
> with 0% CPU load on the relay.
> 
> So the only thing I can think of causing something like what you describe 
> (even though I'm still not sure what you meant by hanging up sessions), is 
> that somehow this process didn't finish setting up completely and the relay 
> directly receives media streams from hundreds of devices because only one 
> endpoint sends data (or the other endpoint's data gets filtered at some 
> firewall), and because it cannot learn both endpoint's addresses it cannot 
> setup the kernel conntrack rule to move data forwarding to the kernel.
> 
> On 14 Mar 2017, at 13:38, Dan Pascu wrote:
> 
> >
> > On 13 Mar 2017, at 18:58, Daniel Zanutti wrote:
> >
> >> Hi guys
> >>
> >> I sent this email a few days ago, anyone from Mediaproxy team could take a 
> >> look? I could debug it, just need some directions on where to look.
> >
> > We have never encountered this problem, so I', not sure what to suggest, 
> > even more considering that the description is not very clear. What do you 
> > mean when you say the relay starts to hang some sessions? That it timeouts 
> > on them not having traffic and initiates a BYE for those sessions? Because 
> > in the next paragraph you imply that they never timeout.
> >
> >>
> >> Thanks
> >>
> >> On Tue, Mar 7, 2017 at 11:10 AM, Daniel Zanutti  >> > wrote:
> >> I'm using mediaproxy on 

Re: [OpenSIPS-Users] rtpengine xmlrpc integration with opensips. Teardown xmlrpc method

[Adrian Georgescu]

> On 29 Jan 2017, at 23:05, Carlos Oliva  wrote:
> 
> Hi List!
> 
> I'm using Opensips 1.11 and doing some tests to change my mediaproxy 
> rtprelays to ngcp-rtpengine. My reasons to try this change are efficiency and 
> that the mediaproxy project seems to be a little abandoned by AGProjects (not 
> really abandoned but has no new features in years)

Well, it is not abandoned. It just works. So there is nothing to fix, hence 
little activity about it.

Adrian


> After the change I started to see some dialogs in state 3 that ends at 
> timeout (6 hours in my config)
> 
> I tried to use the RTPTimeout function in rtpengine (in mediaproxy  it works 
> very well) to try to end the dialogs which don't have RTP.
> 
> To try this I used the rtpengine flags --b2b-url=http://%%:8000/RPC2 
> --xmlrpc-format=1 to send Opensips the order to end the related dialog.
> 
> It doesn't work. Doing some ngrep at xmlrpc interface seems that rtpengine 
> send some commands to opensips RPC interface that Opensips does not 
> understand. The command is: "teardown" and the callid, here is an example:
> 
> POST /RPC2 HTTP/1.1..Host: XXX.XXX.XXX.XXX:8000..Accept: */*..Content-Type: 
> text/xml..User-Agent: Xmlrpc-c/1.33.14 Curl/7.38.0. 
> .Content-Length: 204   
> ding="UTF-8"?>teardown822048991-4075...@bjc.bgi.b.ge
>  
> 
> 
> and the opensips response:
> 
> HTTP/1.1 200 OK..Connection: Keep-Alive..Content-Length: 48..Content-Type: 
> text/xml; charset=utf-8..Date: Sun, 29 Jan 2017 20:31:36 
> GMTInternal server error!
> 
> Obviously OpenSips does not implement this "teardown" method.
> 
> My questions are:
> 
> Anybody has a good idea of how to deal with this?
> 
> Devels: Do you think is a good idea to open a feature request in github about 
> this? I'll try to backport to 1.11 later.if you accept the request.
> 
> 
> Thanks and Regards,
> 
> Carlos Oliva
> 
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Re: [OpenSIPS-Users] iOS 10 Push Notifications

[Adrian Georgescu]

When the Registration expires or it is closed when the device goes to 
background, there is no 408.

The idea is to attempt to wake up the device when call comes in, wait for it to 
register and then fork the original INVITE. Then timeout if no device registers.

Adrian


> On 5 Feb 2017, at 12:53, Aron Podrigal  wrote:
> 
> you can certainly handle this in a t_on_failure for a 408...
> 
> On Thu, Feb 2, 2017, 3:38 PM Adrian Georgescu  <mailto:a...@ag-projects.com>> wrote:
> There is more plumbing needed to replay transactions later and the future 
> OpenSIPS version announced recently will help make this happen.
> 
> You are not alone fighting with this problem.
> 
> Regards,
> Adrian
> 
> 
>> On 1 Feb 2017, at 19:46, Андрей Журавлёв > <mailto:andrei.zhurav...@m-st.ru>> wrote:
>> 
>> Hi All,
>> 
>> I am pretty new with SIP and OpenSIPS. I have problem that affect most of 
>> the people who should support mobile clients for iOS 10. There must be a way 
>> to solve it, but it looks like I missed it. So I need your help.
>> 
>> Background: As you probably already know in iOS 10 Apple prevented network 
>> connections in background mode. It mostly affected VoIP apps. In order to 
>> solve this issue they introduced so-called VoIP Push notifications (via 
>> PushKit) which should automatically wake-up application and allow it to do a 
>> registration and receive a call.
>> 
>> Btw, the only thing I found capable to send VoIP push notifications without 
>> issus is ruby gem/binary called Houston.
>> 
>> Now everything works fine, except one issue actually, when application wakes 
>> up and do a registration, it obviously missed initial INVITE message for a 
>> call, and it looks like server do not retry invites if no provisional 
>> messages returned from a client.
>> 
>> So the question is it possible to tell OpenSIPS server to re-send INVITE 
>> messages (by some timer probably) if no provisional information received 
>> from a client.
>> 
>> I know there is fr_timer and fr_inv_timer params from tm module, but it 
>> looks like they did not do the trick.
>> 
>> Actually, I have almost default config file, except the parts, required for 
>> push notifications.
>> I've posted it here: http://pastebin.com/tZmP320g 
>> <http://pastebin.com/tZmP320g>
>> 
>> Yours sincerely,
>> Andrei Zhuravlev
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>> <http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
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Re: [OpenSIPS-Users] iOS 10 Push Notifications

[Adrian Georgescu]

There is more plumbing needed to replay transactions later and the future 
OpenSIPS version announced recently will help make this happen.

You are not alone fighting with this problem.

Regards,
Adrian


> On 1 Feb 2017, at 19:46, Андрей Журавлёв  wrote:
> 
> Hi All,
> 
> I am pretty new with SIP and OpenSIPS. I have problem that affect most of the 
> people who should support mobile clients for iOS 10. There must be a way to 
> solve it, but it looks like I missed it. So I need your help.
> 
> Background: As you probably already know in iOS 10 Apple prevented network 
> connections in background mode. It mostly affected VoIP apps. In order to 
> solve this issue they introduced so-called VoIP Push notifications (via 
> PushKit) which should automatically wake-up application and allow it to do a 
> registration and receive a call.
> 
> Btw, the only thing I found capable to send VoIP push notifications without 
> issus is ruby gem/binary called Houston.
> 
> Now everything works fine, except one issue actually, when application wakes 
> up and do a registration, it obviously missed initial INVITE message for a 
> call, and it looks like server do not retry invites if no provisional 
> messages returned from a client.
> 
> So the question is it possible to tell OpenSIPS server to re-send INVITE 
> messages (by some timer probably) if no provisional information received from 
> a client.
> 
> I know there is fr_timer and fr_inv_timer params from tm module, but it looks 
> like they did not do the trick.
> 
> Actually, I have almost default config file, except the parts, required for 
> push notifications.
> I've posted it here: http://pastebin.com/tZmP320g 
> 
> 
> Yours sincerely,
> Andrei Zhuravlev
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Re: [OpenSIPS-Users] How to overcome SIP ALG on Wi-Fi routers

[Adrian Georgescu]

+1

> On 02 May 2016, at 21:59, Russell Treleaven  wrote:
> 
> TCP works for me.
> On May 2, 2016 8:43 PM, "Nabeel"  > wrote:
> Thanks for the suggestions of using TLS or changing the port. I changed the 
> port, but some routers are still able to mess with the SIP headers. I would 
> have used TLS, if not for two reasons:
> 
> 1. ICE protocol was originally designed for UDP according to RFC5245, and it 
> seems to work better with UDP.
> 
> 2. The SIP servers I have used (OpenSIPS and Repro) seem to be more stable 
> with UDP compared to TLS (they do not randomly drop connections, throw 
> unusual errors in the logs, etc.)
> 
> I may try TLS again, but it would be better if there is an alternative 
> workaround for UDP.
> 
> On 2 May 2016 at 13:33, Patrick Wakano  > wrote:
> Using TLS!
> Also configuring your systems/devices to use other port than 5060 may do the 
> trick...
> 
> On Mon, May 2, 2016 at 9:14 AM, Nabeel  > wrote:
> Hi,
> 
> Other than using rtpproxy/NAThelper modules, is there any way to 
> bypass/workaround SIP ALG enabled on many WiFi routers? Although SIP ALG was 
> designed to help with NAT, in most cases it does the opposite and breaks SIP.
> 
> Nabeel
> 
> 
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> 
> 
> 
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Re: [OpenSIPS-Users] Register to Cisco call manager on behalf of clients

[Adrian Georgescu]

If you get rid of CUCM, your life would be much much easier.

Adrian


> On 25 Feb 2016, at 05:11, Chen-Che Huang  wrote:
> 
> Dear all,
> 
> We are attempting to use to OpenSIPS to connect to Cisco unified call
> manager (abbreviated as CUCM).  The architecture looks like ``clients <->
> OpenSIPS outbound proxy <-> CUCM''
> Clients register and make calls to CUCM via OpenSIPS outbound proxy. Due to
> the limitation of CUCM, for each client, the OpenSIPS outbound proxy needs
> to use a separate port to register to CUCM (CUCM does not allow different
> clients to register using the same IP:port).
> 
> To address this issue, we would like to modify the source code of OpenSIPS
> (we know the license issue and will release the modified code if we
> accomplish it). The idea of modification is that we would create processes
> inside OpenSIPS and write some APIs to interact with the main processes. For
> example, when the main process finished handling a request, it will use the
> API to make the request be sent to the CUCM by our created process; when
> receiving the response from CUCM, our created process to use the API to
> delegate the main process to handle the response.
> 
> In short, we hope that our created processes won't affect the operations of
> the main process while achieving our goal. I am seeking any comments on this
> issue. Any response is greatly appreciated.
> 
> Best regards,
> Chen-Che
> 
> 
> 
> --
> View this message in context: 
> http://opensips-open-sip-server.1449251.n2.nabble.com/Register-to-Cisco-call-manager-on-behalf-of-clients-tp7601683.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
> 
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Re: [OpenSIPS-Users] New OpenXCAP release 2.4.0

[Adrian Georgescu]

On Debian and Ubuntu you can just do apt-get install openxcap from AGP repos. 
If you build related dependencies from scratch you are here not in the right 
place as this has nothing to do with OpenSIPS or OpenXCAP themselves.

Adrian

> On 28 Feb 2016, at 23:15, sevpal  wrote:
> 
> Hi, there is a message "No Module named sipsimple.core" when tried to start 
> the OpenXCAP server. How can this be resolved? I'm thinking it is referring 
> to the "python-sipsimple" package but that is a beast to compile along with 
> ffmpeg.
> 
> 
> -Original Message- From: Adrian Georgescu
> Sent: Thursday, January 07, 2016 1:06 PM
> To: OpenSIPS users mailling list ; OpenSIPS devel mailling list
> Subject: [OpenSIPS-Users] New OpenXCAP release 2.4.0
> 
> Hello,
> 
> There is a new release of OpenXCAP with various bug fixes
> 
> openxcap (2.4.0) unstable; urgency=medium
> 
> * Removed runtime dependency check
> * Updated copyright years
> 
> openxcap (2.3.0) unstable; urgency=medium
> 
> * Refactor xcapdiff publishing for OpenSIPS backend
> * Fix dependency name
> * Drop dependency on twisted-web
> * Raise python-application version dependency
> * Add dependency on python-sipsimple
> 
> http://openxcap.org
> 
> Regards,
> Adrian
> 
> 
> 
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Re: [OpenSIPS-Users] Linphone and OpenSIPS over TCP

[Adrian Georgescu]

OpenSIPS does not need to interact in anyway with a TURN server. A TURN server 
is used by the SIP client that has built-in TURN functionality. While they may 
run side by side, the only interaction of a sip proxy and TURN server is 
sharing the same database with credentials in order to authenticate the 
end-users.

What MediaProxy does, it simulates a TURN server candidate by inserting it into 
an offer generated by a SIP client that has ICE support. But it is not real 
TURN in the sense that it does not implement the TURN protocol (which does a 
lot of other things, not just the relaying packets over UDP part), and the 
client is unaware of this TURN candidate insertion. Practically, MediaProxy 
only relays RTP packets over UDP and is fouling both end-points into believing 
that during the ICE negotiation there is a relay server that can be used when 
end-to-end RTP does not work. One cannot get RTP over TCP running with this 
hybrid model.

If you want TCP, you need a real TURN client with a real TURN server. 

Regards,
Adrian


> On 06 Feb 2016, at 01:07, Nabeel  wrote:
> 
> On 3 February 2016 at 23:42, sevpal mailto:sev...@aol.com>> 
> wrote:
> 
> Opensips interacts with the TURN in server MediaProxy only.
> 
> 
> That's not completely true, because the TURN server works with OpenSIPS when 
> using UDP for calls. It just doesn't work - or only partially works - with 
> TCP/TLS.  I tested by changing configurations in the TURN server to use TCP 
> relays/listeners only, but it still only works with UDP. This leads me to 
> believe that it is a limitation in OpenSIPS that it only handles the TURN 
> server properly when calls are using UDP, not TCP/TLS.
> 
> Coturn is probably the most comprehensive TURN server available today. Is 
> there any chance that OpenSIPS will fully support this in the future?
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Re: [OpenSIPS-Users] REG::VOIP Client with TLS Support

[Adrian Georgescu]

Try Blink

http://icanblink.com 


> On 12 Nov 2015, at 12:53, Ravitez Ravi  wrote:
> 
> Hello All,
> Can you please suggest a Free VOIP client with TLS support.
> Thank you for your time.
> 
> Regards,
> Ravitez.D
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Re: [OpenSIPS-Users] Does Media_Relay component implies in impossible direct media? (Sorry)

[Adrian Georgescu]

> On 20 Oct 2015, at 10:20, Rodrigo Pimenta Carvalho  wrote:
> 
> Hi Adrian.
> 
> Sorry any inconvenience from my poor interpretation. I'm completely new to 
> such NAT subjects.
> 
> I will consider:
> 
> If I don’t use STUN and when using ICE + MediaProxy, media will go directly 
> between end-points behind the same NAT. The rest of the calls will go through 
> media relay.
> 
> So, MediaProxy is not a substitute for a STUN server.
> 
> In this case, as I need direct media, I will give up of using MediaProxy and 
> use just STUN technology.

STUN does not work, a large percentage of calls will fail by using it alone 
without a media relay. These things are complementary and you must use them all 
to cover 100% of the cases.

> 
> Hopefully, now I got the point!
> 
> Regards.
> 
> 
> 
> RODRIGO PIMENTA CARVALHO
> Inatel Competence Center
> Software
> Ph: +55 35 3471 9200 RAMAL 979
> 
> 
> De: users-boun...@lists.opensips.org 
> <mailto:users-boun...@lists.opensips.org>  <mailto:users-boun...@lists.opensips.org>> em nome de Adrian Georgescu 
> mailto:a...@ag-projects.com>>
> Enviado: terça-feira, 20 de outubro de 2015 10:47
> Para: OpenSIPS users mailling list
> Assunto: Re: [OpenSIPS-Users] Does Media_Relay component implies in 
> impossible direct media?
>  
> Read again what I wrote:
> 
> If you don’t use STUN and when using ICE, media will go directly between 
> end-points behind the same NAT. The rest of the calls will go through media 
> relay.
> 
> 
>> On 20 Oct 2015, at 07:48, Rodrigo Pimenta Carvalho > <mailto:pime...@inatel.br>> wrote:
>> 
>> Ok Adrian.
>> 
>> Thank you too!
>> 
>> But I am still trying to find the answer for the following question, before 
>> deciding to install MediaProxy:
>> 
>> Can MediaProxy, by itself, without a separated STUN server, provide all 
>> necessary functionalities to my system get direct media between UACs?
>> 
>> 
>> 
>> Best regards.
>> 
>> 
>> RODRIGO PIMENTA CARVALHO
>> Inatel Competence Center
>> Software
>> Ph: +55 35 3471 9200 RAMAL 979
>> 
>> 
>> De: users-boun...@lists.opensips.org 
>> <mailto:users-boun...@lists.opensips.org> > <mailto:users-boun...@lists.opensips.org>> em nome de Adrian Georgescu 
>> mailto:a...@ag-projects.com>>
>> Enviado: segunda-feira, 19 de outubro de 2015 18:54
>> Para: OpenSIPS users mailling list
>> Assunto: Re: [OpenSIPS-Users] Does Media_Relay component implies in 
>> impossible direct media?
>>  
>> If you don’t use STUN and when using ICE, media will go directly between 
>> end-points behind the same NAT. The rest of the calls will go through media 
>> relay.
>> 
>> Adrian
>> 
>>> On 19 Oct 2015, at 11:03, Nabeel >> <mailto:nabeelshik...@gmail.com>> wrote:
>>> 
>>> ICE in general uses STUN (RFC 5245), however I do not use Mediaproxy myself 
>>> so cannot comment on how it handles STUN.   Someone with Mediaproxy may be 
>>> able to answer.
>>> On 19 Oct 2015 14:41, "Rodrigo Pimenta Carvalho" >> <mailto:pime...@inatel.br>> wrote:
>>> Ok Nabeel.
>>> 
>>> In my environment, I intend to have MediaProxy and all clients supporting 
>>> ICE.
>>> However, in my environment/network doesn't exist a STUN server.
>>> 
>>> So, my last questions (before I start the MediaProxy installation 
>>> procedures) about this subject are:
>>> 
>>> 1 -  Does MediaProxy work as a STUN server, or just as a TURN server?
>>> 
>>> 2 - Is it always necessary to provide a STUN server separately, even using 
>>> the MediaProxy, if direct media is desired?
>>> 
>>> 3 - Let's suppose I will not provide a separately STUN server, in this case 
>>> will the media always pass through the media relay,?
>>> 
>>> Thank you again!
>>> Best regards.
>>> 
>>> 
>>> RODRIGO PIMENTA CARVALHO
>>> Inatel Competence Center
>>> Software
>>> Ph: +55 35 3471 9200  RAMAL 979
>>> 
>>> 
>>> De: users-boun...@lists.opensips.org 
>>> <mailto:users-boun...@lists.opensips.org> >> <mailto:users-boun...@lists.opensips.org>> em nome de Nabeel 
>>> mailto:nabeelshik...@gmail.com>>
>>> Enviado: segunda-feira, 19 de outubro de 2015 10:40
>>> Para: OpenSIPS users mailling list
>>> Assunto: Re: [OpenSIPS-Users] Does Media_Relay component implies in 
&

Re: [OpenSIPS-Users] Does Media_Relay component implies in impossible direct media?

[Adrian Georgescu]

Read again what I wrote:

If you don’t use STUN and when using ICE, media will go directly between 
end-points behind the same NAT. The rest of the calls will go through media 
relay.


> On 20 Oct 2015, at 07:48, Rodrigo Pimenta Carvalho  wrote:
> 
> Ok Adrian.
> 
> Thank you too!
> 
> But I am still trying to find the answer for the following question, before 
> deciding to install MediaProxy:
> 
> Can MediaProxy, by itself, without a separated STUN server, provide all 
> necessary functionalities to my system get direct media between UACs?
> 
> 
> 
> Best regards.
> 
> 
> RODRIGO PIMENTA CARVALHO
> Inatel Competence Center
> Software
> Ph: +55 35 3471 9200 RAMAL 979
> 
> 
> De: users-boun...@lists.opensips.org 
> <mailto:users-boun...@lists.opensips.org>  <mailto:users-boun...@lists.opensips.org>> em nome de Adrian Georgescu 
> mailto:a...@ag-projects.com>>
> Enviado: segunda-feira, 19 de outubro de 2015 18:54
> Para: OpenSIPS users mailling list
> Assunto: Re: [OpenSIPS-Users] Does Media_Relay component implies in 
> impossible direct media?
>  
> If you don’t use STUN and when using ICE, media will go directly between 
> end-points behind the same NAT. The rest of the calls will go through media 
> relay.
> 
> Adrian
> 
>> On 19 Oct 2015, at 11:03, Nabeel > <mailto:nabeelshik...@gmail.com>> wrote:
>> 
>> ICE in general uses STUN (RFC 5245), however I do not use Mediaproxy myself 
>> so cannot comment on how it handles STUN.   Someone with Mediaproxy may be 
>> able to answer.
>> On 19 Oct 2015 14:41, "Rodrigo Pimenta Carvalho" > <mailto:pime...@inatel.br>> wrote:
>> Ok Nabeel.
>> 
>> In my environment, I intend to have MediaProxy and all clients supporting 
>> ICE.
>> However, in my environment/network doesn't exist a STUN server.
>> 
>> So, my last questions (before I start the MediaProxy installation 
>> procedures) about this subject are:
>> 
>> 1 -  Does MediaProxy work as a STUN server, or just as a TURN server?
>> 
>> 2 - Is it always necessary to provide a STUN server separately, even using 
>> the MediaProxy, if direct media is desired?
>> 
>> 3 - Let's suppose I will not provide a separately STUN server, in this case 
>> will the media always pass through the media relay,?
>> 
>> Thank you again!
>> Best regards.
>> 
>> 
>> RODRIGO PIMENTA CARVALHO
>> Inatel Competence Center
>> Software
>> Ph: +55 35 3471 9200  RAMAL 979
>> 
>> 
>> De: users-boun...@lists.opensips.org 
>> <mailto:users-boun...@lists.opensips.org> > <mailto:users-boun...@lists.opensips.org>> em nome de Nabeel 
>> mailto:nabeelshik...@gmail.com>>
>> Enviado: segunda-feira, 19 de outubro de 2015 10:40
>> Para: OpenSIPS users mailling list
>> Assunto: Re: [OpenSIPS-Users] Does Media_Relay component implies in 
>> impossible direct media?
>>  
>> Yes, that is how it should work.  ICE tries to connect directly first before 
>> trying other methods.  If direct connection fails, then it tries STUN, then 
>> media relay.  The clients have to support ICE too.
>> 
>> On 19 October 2015 at 11:35, Rodrigo Pimenta Carvalho > <mailto:pime...@inatel.br>> wrote:
>> Hi Nabeel.
>> 
>> Thank you very much for your answer.
>> If I understood it well, while using MediaProxy, the clients may connect 
>> directly, but they may not too. ICE will decide it for me and I don't need 
>> to worry about it.
>> Tell me if I got it, please.
>> 
>> Best regards.
>> 
>> 
>> 
>> RODRIGO PIMENTA CARVALHO
>> Inatel Competence Center
>> Software
>> Ph: +55 35 3471 9200  RAMAL 979
>> 
>> 
>> De: users-boun...@lists.opensips.org 
>> <mailto:users-boun...@lists.opensips.org> > <mailto:users-boun...@lists.opensips.org>> em nome de Nabeel 
>> mailto:nabeelshik...@gmail.com>>
>> Enviado: sábado, 17 de outubro de 2015 19:56
>> Para: OpenSIPS users mailling list
>> Assunto: Re: [OpenSIPS-Users] Does Media_Relay component implies in 
>> impossible direct media?
>>  
>> Rodrigo,
>> It is the responsibility of ICE to determine in which situation it is best 
>> to establish a direct connection, versus use of STUN, versus use of a media 
>> relay.
>> We cannot manually decide which clients will connect directly and which will 
>> not.
>> On 17 Oct 2015 22:07, "Rodrigo Pimenta Carvalho" > <mailto:pime...@inatel.br>> wrote:
>> 
>> Hi.
>&g

Re: [OpenSIPS-Users] Does Media_Relay component implies in impossible direct media?

[Adrian Georgescu]

If you don’t use STUN and when using ICE, media will go directly between 
end-points behind the same NAT. The rest of the calls will go through media 
relay.

Adrian

> On 19 Oct 2015, at 11:03, Nabeel  wrote:
> 
> ICE in general uses STUN (RFC 5245), however I do not use Mediaproxy myself 
> so cannot comment on how it handles STUN.   Someone with Mediaproxy may be 
> able to answer.
> 
> On 19 Oct 2015 14:41, "Rodrigo Pimenta Carvalho"  > wrote:
> Ok Nabeel.
> 
> 
> In my environment, I intend to have MediaProxy and all clients supporting ICE.
> 
> However, in my environment/network doesn't exist a STUN server.
> 
> 
> So, my last questions (before I start the MediaProxy installation procedures) 
> about this subject are:
> 
> 
> 1 -  Does MediaProxy work as a STUN server, or just as a TURN server?
> 
> 
> 2 - Is it always necessary to provide a STUN server separately, even using 
> the MediaProxy, if direct media is desired?
> 
> 
> 3 - Let's suppose I will not provide a separately STUN server, in this case 
> will the media always pass through the media relay,?
> 
> 
> Thank you again!
> 
> Best regards.
> 
> 
> RODRIGO PIMENTA CARVALHO
> Inatel Competence Center
> Software
> Ph: +55 35 3471 9200  RAMAL 979
> 
> 
> De: users-boun...@lists.opensips.org 
>   > em nome de Nabeel 
> mailto:nabeelshik...@gmail.com>>
> Enviado: segunda-feira, 19 de outubro de 2015 10:40
> Para: OpenSIPS users mailling list
> Assunto: Re: [OpenSIPS-Users] Does Media_Relay component implies in 
> impossible direct media?
>  
> Yes, that is how it should work.  ICE tries to connect directly first before 
> trying other methods.  If direct connection fails, then it tries STUN, then 
> media relay.  The clients have to support ICE too.
> 
> On 19 October 2015 at 11:35, Rodrigo Pimenta Carvalho  > wrote:
> Hi Nabeel.
> 
> 
> Thank you very much for your answer.
> 
> If I understood it well, while using MediaProxy, the clients may connect 
> directly, but they may not too. ICE will decide it for me and I don't need to 
> worry about it.
> 
> Tell me if I got it, please.
> 
> 
> Best regards.
> 
> 
> 
> RODRIGO PIMENTA CARVALHO
> Inatel Competence Center
> Software
> Ph: +55 35 3471 9200  RAMAL 979
> 
> 
> De: users-boun...@lists.opensips.org 
>   > em nome de Nabeel 
> mailto:nabeelshik...@gmail.com>>
> Enviado: sábado, 17 de outubro de 2015 19:56
> Para: OpenSIPS users mailling list
> Assunto: Re: [OpenSIPS-Users] Does Media_Relay component implies in 
> impossible direct media?
>  
> Rodrigo,
> 
> It is the responsibility of ICE to determine in which situation it is best to 
> establish a direct connection, versus use of STUN, versus use of a media 
> relay.
> 
> We cannot manually decide which clients will connect directly and which will 
> not.
> 
> On 17 Oct 2015 22:07, "Rodrigo Pimenta Carvalho"  > wrote:
> 
> Hi.
> 
> 
> In my project, UACs should run direct media. RTP packets direct transmitted 
> from each other.
> 
> However, due to NAT presence, I'm thinking about to use MediaProxy. 
> 
> 
> In this case, is it possible to get direct media after some UAC negotiations 
> passing through the Media_Relay component, or is it impossible? That is, does 
> the media will always pass through the Media_Relay host too?
> 
> 
> Any hint will be very helpful!
> 
> Thanks alot.
> 
> 
> 
> 
> RODRIGO PIMENTA CARVALHO
> Inatel Competence Center
> Software
> Ph: +55 35 3471 9200  RAMAL 979
> 
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Re: [OpenSIPS-Users] Does Media_Relay component implies in impossible direct media?

[Adrian Georgescu]

Yes if both clients have support for ICE standard.

> On 17 Oct 2015, at 18:06, Rodrigo Pimenta Carvalho  wrote:
> 
> 
> Hi.
> 
> In my project, UACs should run direct media. RTP packets direct transmitted 
> from each other.
> However, due to NAT presence, I'm thinking about to use MediaProxy.
> 
> In this case, is it possible to get direct media after some UAC negotiations 
> passing through the Media_Relay component, or is it impossible? That is, does 
> the media will always pass through the Media_Relay host too?
> 
> Any hint will be very helpful!
> Thanks alot.
> 
> 
> 
> 
> RODRIGO PIMENTA CARVALHO
> Inatel Competence Center
> Software
> Ph: +55 35 3471 9200 RAMAL 979
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Re: [OpenSIPS-Users] What choise? MediaProxy module or RTPProxy module?

[Adrian Georgescu]

> On 15 Oct 2015, at 14:39, Eric Tamme  wrote:
> 
> Mediaproxy and rtpproxy serve identical purposes.  They both can act as a 
> relay to help relay media between clients on different NAT's.  If you dont 
> want to relay media when clients are behind the same nat - you must inspect 
> the SDP on your proxy and chose not to use a media relay.
> 
> If you have clients behind 2 different NAT's it is generally not possible to 
> pass media without using a relay server.

Well not really, is very likely to get the media through by using ICE in both 
clients even both are behind NAT. Only when both clients are behind symmetric 
NAT, which is just one scenario out of many, your statement is true.

Adrian

> 
> -Eric
> 
> On 10/15/2015 11:36 AM, Rodrigo Pimenta Carvalho wrote:
>> 
>> Hi.
>> 
>> Today I'm searching for a solution that allows me to use OpenSIPS, SIP over 
>> TCP, end-nodes behind NATs and direct media.
>> 
>> As someone pointed, I'm reading now about MediaProxy module.
>> 
>> However, if I'm right, by using MediaProxy I will get only media relay. That 
>> is, MediaProxy does only provide media relay. Am I right?
>> 
>> I need an way to implement direct media between the end-nodes, behind NATs. 
>> I want to avoid pass the media through the MediaProxy, due to performance 
>> reasons.
>> 
>> Should I give up using MediaProxy and move to RTPProxy? Would it be a good 
>> decision?
>> 
>> When it is better to use MediaProxy, and when it is better to use RTPProxy? 
>> 
>> Any hint will be very helpful!
>> 
>> Best regards.
>> 
>> 
>> RODRIGO PIMENTA CARVALHO
>> Inatel Competence Center
>> Software
>> Ph: +55 35 3471 9200 RAMAL 979
>> 
>> 
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Re: [OpenSIPS-Users] What to choose: Nat Traversal, Media Proxy or Nathelper ?? I need a direction.

[Adrian Georgescu]

You need MediaProxy with ICE mode enabled.

Adrian

> On 15 Oct 2015, at 09:47, Rodrigo Pimenta Carvalho  wrote:
> 
> Dear OpenSIPS-Users,
> 
> I'm working in a project that needs to use OpensSIPS, SIP over TCP,  
> softphones behind NAT, and video transmission between the sip proxy and the 
> softphones as direct media.
> 
> As I'm completely new to NAT traversing and direct media, I wonder what is 
> the first module to study?
> 
> Nat Traversal, or Media Proxy, or Nathelper?
> 
> What of these 3 modules will be really necessary, to provide direct media, 
> when using SIP over IP and peers behind NATs?
> 
> Any hint will be very helpful!
> 
> Regards.
> 
> RODRIGO PIMENTA CARVALHO
> Inatel Competence Center
> Software
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Re: [OpenSIPS-Users] Presence Server performance test

[Adrian Georgescu]

You need to take handling of XCAP policy documents into account.

Adrian

> On 30 Sep 2015, at 20:57, surya  wrote:
> 
> Hi, 
> 
> I want to test the performance of presence server in different scenarios.
> How can I do that properly? 
> 
> I know the tool is SIPP for generating load and tested few scenarios like
> PUBLISH and SUBSCRIBE-NOTIFY. But not sure of the proper way in terms of
> what all parameters to set while sending the sipp request and the scenarios
> to test. 
> 
> Example Scenarios
> 1. PUBLISH (new with no subscriber) 
> 2. SUBSCRIBE - NOTIFY 
> 3. PUBLISH - NOTIFY (refresh) 
> What else? 
> 
> Now the sipp parameters, I am confused what values to set for rate, period
> and max or how to vary them. 
> 
> I have never done performance testing so basically what I need is some
> guidance to do some tests which makes some sense.
> 
> Thanks, 
> Surya
> 
> 
> 
> --
> View this message in context: 
> http://opensips-open-sip-server.1449251.n2.nabble.com/Presence-Server-performance-test-tp7599299.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
> 
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Re: [OpenSIPS-Users] force_tcp_alias default behavior seems wrong – should not use port from Via-header

[Adrian Georgescu]

> On 23 Sep 2015, at 22:24, Jonas Borjesson  wrote:
> 
> Hey,
> 
> Thanks for you reply. Just some additional notes I perhaps should have
> included. How the client arrives to which Contact address it has isn't
> the main issue here, opensips can fix that easily (e.g.
> fix_nated_address) and in fact, these clients don't actually use STUN
> per se, they use the information in the Via of the response coming
> back (recieved & rport) to "fix" their contact and will re-register.
> However, the scenario I describe will still happen for clients behind
> the very same NAT due to how the force_tcp_alias works and currently
> it will grab the port off of the via, which, as you also argues, is
> unsafe and you should rely on information from the received packet
> itself. So, it sounds like we are arguing pretty much the same thing,
> i.e., don't trust the client :-)
> 
> So, your statement
> 
> "Two end-points behind same NAT will always be different from the
> server point of view as there will be different ports opened in the
> NAT-ed router. Starting with wrong assumptions like the ones above
> one can easily end up in the wrong places to look for solutions."
> 
> is correct except that, again, force_tcp_alias will not allow you to
> map those two different connections to two different entries within
> opensips. Does that make sense?

I am not an expert in this particular issue. This is why I wrote that "below is 
for others to comment”.

> 
> /Jonas
> 
> 
> On Wed, Sep 23, 2015 at 6:05 PM, Adrian Georgescu  
> wrote:
>> 
>>> On 23 Sep 2015, at 21:34, Jonas Borjesson  wrote:
>>> 
>>> Hi all,
>>> 
>>> I have the following problem (sorry for somewhat long explanation,
>>> want to get it right on the first try :-):
>>> 
>>> * Alice is behind a NAT and registers with whatever.com and has
>>> through STUN figured out her public ip, which will go into the Contact
>>> of the REGISTER.
>> 
>> This is unreliable, you should never use STUN for this purpose. A well 
>> configured server will never use the information presented by an end-point 
>> in the Contact header of a Register, it will use the actual transport and 
>> port of the received packet instead.
>> 
>>> * Bob is behind the same NAT as Alice and registers with whatever.com
>>> and has through STUN figured out his public ip, which will go into the
>>> Contact of the REGISTER.
>> 
>> Again, this is unreliable, you should never use STUN for this purpose. 
>> OpenSIPS is smart enough to determine where the end-point registered from 
>> instead of trusting the information in the Contact header which can be 
>> wrong, fake or anything in between.
>> 
>> Two end-points behind same NAT will always be different from the server 
>> point of view as there will be different ports opened in the NAT-ed router. 
>> Starting with wrong assumptions like the ones above  one can easily end up 
>> in the wrong places to look for solutions.
>> 
>> Below is for others to comment.
>> 
>>> * Carol is another user that calls b...@whatever.com.
>>> * opensips is acting as a pure transaction stateful proxy for all SIP
>>> traffic (including REGISTER so I'm not using opensips as a registrar)
>>> * opensips config is using force_tcp_alias() so that the connection
>>> can be re-used at a later point.
>>> * Alice's and Bob's clients are sending keep-alive traffic (double
>>> crlf) to keep the connection up.
>>> 
>>> In the above scenario, because of the way force_tcp_alias works, Bob
>>> will NOT get the call but rather Alice for the following reason:
>>> 
>>> When Alice's client registers and the force_tcp_alias is executed, a
>>> mapping between Alice's public ip + the port found in the top-most
>>> Via-header and her incoming TCP/TLS connection will be created. When
>>> Bob registers, which he does after Alice, he too will create mapping
>>> between his connection and the public_ip + port in Via. Of course,
>>> since Alice and Bob are behind the same NAT they will have the same
>>> IP, hence “half” the key is shared between Alice & Bob at all times.
>>> If Alice & Bob also puts the same port as each other in the top-most
>>> Via-header, they now share the exact same key for the connection
>>> whereby opensips will complain with the following message:
>>> “tcpconn_add_alias: possible port hijack attempt”. Hence, when Carol
>>> later on calls Bob, which then will be “resolved” 

Re: [OpenSIPS-Users] force_tcp_alias default behavior seems wrong – should not use port from Via-header

[Adrian Georgescu]

> On 23 Sep 2015, at 21:34, Jonas Borjesson  wrote:
> 
> Hi all,
> 
> I have the following problem (sorry for somewhat long explanation,
> want to get it right on the first try :-):
> 
> * Alice is behind a NAT and registers with whatever.com and has
> through STUN figured out her public ip, which will go into the Contact
> of the REGISTER.

This is unreliable, you should never use STUN for this purpose. A well 
configured server will never use the information presented by an end-point in 
the Contact header of a Register, it will use the actual transport and port of 
the received packet instead.

> * Bob is behind the same NAT as Alice and registers with whatever.com
> and has through STUN figured out his public ip, which will go into the
> Contact of the REGISTER.

Again, this is unreliable, you should never use STUN for this purpose. OpenSIPS 
is smart enough to determine where the end-point registered from instead of 
trusting the information in the Contact header which can be wrong, fake or 
anything in between.

Two end-points behind same NAT will always be different from the server point 
of view as there will be different ports opened in the NAT-ed router. Starting 
with wrong assumptions like the ones above  one can easily end up in the wrong 
places to look for solutions.

Below is for others to comment.

> * Carol is another user that calls b...@whatever.com.
> * opensips is acting as a pure transaction stateful proxy for all SIP
> traffic (including REGISTER so I'm not using opensips as a registrar)
> * opensips config is using force_tcp_alias() so that the connection
> can be re-used at a later point.
> * Alice's and Bob's clients are sending keep-alive traffic (double
> crlf) to keep the connection up.
> 
> In the above scenario, because of the way force_tcp_alias works, Bob
> will NOT get the call but rather Alice for the following reason:
> 
> When Alice's client registers and the force_tcp_alias is executed, a
> mapping between Alice's public ip + the port found in the top-most
> Via-header and her incoming TCP/TLS connection will be created. When
> Bob registers, which he does after Alice, he too will create mapping
> between his connection and the public_ip + port in Via. Of course,
> since Alice and Bob are behind the same NAT they will have the same
> IP, hence “half” the key is shared between Alice & Bob at all times.
> If Alice & Bob also puts the same port as each other in the top-most
> Via-header, they now share the exact same key for the connection
> whereby opensips will complain with the following message:
> “tcpconn_add_alias: possible port hijack attempt”. Hence, when Carol
> later on calls Bob, which then will be “resolved” to
> bob@public_ip:some_port (by a location aware proxy behind opensips)
> and subsequently proxied to Bob via the opensips node, it will find a
> live connection and re-use that, which ends up at Alice.
> 
> So, looking at the code (action.c):
> 
> case FORCE_TCP_ALIAS_T:
>…
>if (a->elem[0].type==NOSUBTYPE) {
>port=msg->via1->port;
>...
> 
> which clearly grabs the port out of the Via. By doing this there is a
> chance that clients will happen to have the same local port and you
> will run into the scenario above. Even worse, for those clients that
> do not set the port on the Via to the actual port of the connection
> (which clients do) they will end up with the default for the
> transport, which in my case was 5061 for TLS. Malicious users behind
> the same corporate NAT could take advantage of this by setting up many
> connections to effectively “steal” other peoples phone calls, granted,
> they may have to setup several thousands of connections to be sure so
> perhaps not practical.
> 
> So, suggested solutions:
> 
> Suggestion 1: I do not see any value with using the port from the via
> but rather always use the port from the src packet itself since that
> is what you really want anyway. That will avoid everything above.
> 
> Suggestion 2: Allow for script variables to force_tcp_alias so you
> could pass whatever you want, which in my case always would be the
> source port of the incoming packet. Currently, the config-file grammar
> only allows for number but could be easily extended to allow for other
> types as well.
> 
> Both solutions are fairly trivial where solution 1 seems to be the
> correct default behavior but solution 2 has the most flexibility and
> also wouldn't mess with any existing deployments in the wild, even
> though I'm guessing they suffer from the same problem as described but
> may not have been discovered yet.
> 
> Comments/thoughts? If people agree, I will issue a pull request
> against latest 1.11. Also, the behavior is the same for at least
> versions 1.8 and 1.11. I am assuming it’s also the same for all
> versions in between as well, and possibly earlier versions but I
> haven't checked.
> 
> Thanks,
> 
> /Jonas
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Re: [OpenSIPS-Users] Migrating from RTPProxy to MediaProxy

[Adrian Georgescu]

> On 10 Sep 2015, at 11:26, Terrance Devor  wrote:
> 
> Hello Everyone,
> 
> Our current NAT'ed environment is as follows
> 
> Internet <---> Cisco Green Box <-> OpenSIPS (192.168.2.2)
>   <-> RTPProxy 
> (192.168.2.5)
> 
> I think this is called far end nat? Anyhow, it works perfectly fine (ie, 
> flows media) however, we are looking to migrate to MediaProxy to take 
> advantage of CDR tool as well as it's ICE implementation.
> Can MediaProxy run in a natted environment such as Amazon AWC etc...? With 
> RTPProxy we can specify Public Private IP address using -l:

Yes, Mediaproxy can run on Amazon platform.

> 
> rtpproxy -s udp:192.168.2.5:7789  -l 
> 192.168.2.5/74.1.1.1.1  -m 8000 -M 65535 -u 
> root root -F -d INFO LOG_LOCAL0
> 
> Is this also possible with MediaProxy? Also, how difficult is it to migrate 
> to MediaProxy from within OpenSIPS?

Is two lines of code to enable the module and use it.

> 
> Kind Regards,
> 
> Terrance
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Re: [OpenSIPS-Users] Using STUN/TURN server with OpenSIPS

[Adrian Georgescu]

A TURN server is used by a client. The SIP server has no interaction with a 
TURN server, you can deploy that separately.

When you use MediaProxy it provides a TURN server candidate and you do not need 
a separate TURN server software.

Adrian  

> On 24 Jul 2015, at 04:25, Nabeel  wrote:
> 
> So OpenSIPS does not support use of a TURN server without installing an 
> external module (Mediaproxy, RTPproxy)?  Is that right?
> 
> On 23 July 2015 at 20:11, Tito Cumpen  > wrote:
> There is a stun module you can use for opensips
> http://www.opensips.org/html/docs/modules/2.2.x/stun.html 
> 
> as far as TURN if you're using ICE a turn relay can be used and added to the 
> sdp as a lower priority candidate or intermediary while ICE is in discovery . 
> I am familiar with media proxy possessing this capability as well as 
> rtpengine. I'd look into them.
> 
> Thanks,
> Tito
> 
> On Thu, Jul 23, 2015 at 2:27 PM, Nabeel  > wrote:
> Hi,
> 
> I have a STUN/TURN server set up in my SIP clients which also support ICE.  
> However, OpenSIPS does not make use of this STUN/TURN server when attempting 
> to make a call.  
> 
> How do I configure OpenSIPS to use the STUN/TURN server at a given port?
> 
> Which is better to use: RTPproxy or STUN/TURN server, and why?
> 
> 
> 
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Re: [OpenSIPS-Users] CDRTool prepaid for big installation

[Adrian Georgescu]

On 14 Oct 2014, at 13:02, Satish Patel  wrote:

> I am reading this document to implement Quota 
> http://cdrtool.ag-projects.com/projects/cdrtool/repository/entry/doc/QuotaSystem.txt
> 
> In number  3. Configure OpenSIPS to deny sessions initiated by subscribers 
> belonging to the quota group.
> 
> what that means? How do i configure opensips to deny session, is it part of 
> callcontrol config? 


Check if the user belongs to that group and if yes don't let the call go 
through. It has nothing to do with call control, is just a database query.

Adrian



> 
> also it is saying belongs to the quota group, what is quota group? this is 
> what i have in my subscriber table. Even after quota exceed it is not 
> blocking call.  
> 
> +-+--+---+--+---+--+--+--+---+
> | id  | username | domain| password | email_address | ha1 
>  | ha1b | rpid | 
> quota |
> +-+--+---+--+---+--+--+--+---+
> |   1 | 3003 | sip.example.com  | 3003 |   | 
> 6fc4d74adfdedf25c72134154d3a9e1a | 3251dd8a5ef5a35d79a358bccb2c8179 | NULL |  
>1 |
> 
> 
> 
> 
> 
> 
> On Sun, Oct 12, 2014 at 7:30 PM, Adrian Georgescu  
> wrote:
> Quota works very simple. It is a mere cronjob that compares the total amount 
> of costs made in the current calendar month with the maximum allowed. If the 
> costs are higher than the quota, then the SIP account is blocked. This will 
> not cut the calls in progress but it works well statistically speaking for 
> postpaid customers. The documentation explains the modus operandi in more 
> detail.
> 
> Adrian
> 
> On 12 Oct 2014, at 11:14, Satish Patel  wrote:
> 
>> Thanks!! I think you got my point, we have very high density call ratio that 
>> is why prepaid not going to be a solution, I think postpaid or quota would 
>> be right one.  
>> 
>> I have following question:
>> 
>> Postpaid:
>> 
>> Default it treat all calls as postpaid but in case i want to give number of 
>> mins or time to my single customer then how do i achieve that  Example:  
>> 5000 mins or say $500 deposit in customer account then how i can do that 
>> with postpaid? 
>> 
>> 
>> Quota: I never  explore this feature so i just want to know how quota system 
>> work with CDRTool? could you give me short explanation? Most of our customer 
>> would be call center or high density call customer, how i can use quota in 
>> that scenario?  
>> 
>> On Sun, Oct 12, 2014 at 9:38 AM, Adrian Georgescu  
>> wrote:
>> 
>> On 12 Oct 2014, at 09:48, Satish Patel  wrote:
>> 
>>> I have run sipp test and it only able to handle 30 calls and later all call 
>>> failed,
>> 
>> Can you explain what exactly failed?
>> 
>>> I heard from other user post, CDRTool prepaid can't handle many 
>>> simultaneous running calls.
>> 
>> You heard wrong. It cannot handle high density call attempts like calls 
>> generated from call centers or transit peers like SIP trunks that push lot 
>> of calls. The number of simultaneous calls is irrelevant. You can have 
>> thousands of simultaneous calls with almost no performance penalty if the 
>> traffic is generated by regular SIP user devices.
>> 
>>> In our case single account will make many simultaneous calls and we need to 
>>> handle them via prepaid.. 
>> 
>> It all depends on the meaning of many. Whenever a new call is attempted, the 
>> maximum remaining time of all ongoing calls of the same user must be 
>> recalculated so that the balance cannot be exceeded for any of them. This 
>> means that the more calls for the same user you have, the longer it takes to 
>> calculated everything over and over again.
>> 
>> If you have many users with a few calls each like in a residential scenario 
>> where a user makes one or perhaps two parallel calls, this would have little 
>> impact as there is little to re-calculate.
>> 
>>> Some one suggested don't use prepaid because of limitation and performance, 
>>> and suggested use Postpaid or Quota system..  is that true?  
>> 
>> It all depends on the traffic patterns. Concurrent or simultaneous calls is 
>> one thing, high density calls/per second attempts is another. There is no 
>> hard limitation but the number of database queries, 

Re: [OpenSIPS-Users] CDRTool prepaid for big installation

[Adrian Georgescu]

Quota works very simple. It is a mere cronjob that compares the total amount of 
costs made in the current calendar month with the maximum allowed. If the costs 
are higher than the quota, then the SIP account is blocked. This will not cut 
the calls in progress but it works well statistically speaking for postpaid 
customers. The documentation explains the modus operandi in more detail.

Adrian

On 12 Oct 2014, at 11:14, Satish Patel  wrote:

> Thanks!! I think you got my point, we have very high density call ratio that 
> is why prepaid not going to be a solution, I think postpaid or quota would be 
> right one.  
> 
> I have following question:
> 
> Postpaid:
> 
> Default it treat all calls as postpaid but in case i want to give number of 
> mins or time to my single customer then how do i achieve that  Example:  5000 
> mins or say $500 deposit in customer account then how i can do that with 
> postpaid? 
> 
> 
> Quota: I never  explore this feature so i just want to know how quota system 
> work with CDRTool? could you give me short explanation? Most of our customer 
> would be call center or high density call customer, how i can use quota in 
> that scenario?  
> 
> On Sun, Oct 12, 2014 at 9:38 AM, Adrian Georgescu  
> wrote:
> 
> On 12 Oct 2014, at 09:48, Satish Patel  wrote:
> 
>> I have run sipp test and it only able to handle 30 calls and later all call 
>> failed,
> 
> Can you explain what exactly failed?
> 
>> I heard from other user post, CDRTool prepaid can't handle many simultaneous 
>> running calls.
> 
> You heard wrong. It cannot handle high density call attempts like calls 
> generated from call centers or transit peers like SIP trunks that push lot of 
> calls. The number of simultaneous calls is irrelevant. You can have thousands 
> of simultaneous calls with almost no performance penalty if the traffic is 
> generated by regular SIP user devices.
> 
>> In our case single account will make many simultaneous calls and we need to 
>> handle them via prepaid.. 
> 
> It all depends on the meaning of many. Whenever a new call is attempted, the 
> maximum remaining time of all ongoing calls of the same user must be 
> recalculated so that the balance cannot be exceeded for any of them. This 
> means that the more calls for the same user you have, the longer it takes to 
> calculated everything over and over again.
> 
> If you have many users with a few calls each like in a residential scenario 
> where a user makes one or perhaps two parallel calls, this would have little 
> impact as there is little to re-calculate.
> 
>> Some one suggested don't use prepaid because of limitation and performance, 
>> and suggested use Postpaid or Quota system..  is that true?  
> 
> It all depends on the traffic patterns. Concurrent or simultaneous calls is 
> one thing, high density calls/per second attempts is another. There is no 
> hard limitation but the number of database queries, distance to MySQL 
> database will affect how many calls you can handle because as I explained 
> before all concurrent calls must be rerated in real time again for each new 
> call attempt. If one SIP account generates 10K parallel calls the load is 
> infinite while if you have 10K users with one call each the load is almost 
> zero.
> 
> This is why a prepaid model is not practical for high density of calls and 
> this has little to do with CDRTool, any other system would face the same 
> problem, the load is compounded when adding more calls for same account. A 
> quota based system is more appropriate for entities that generate large 
> amount of calls as nothing has to be calculated on a per call basis.
> 
> Adrian
> 
>> On Thu, Oct 9, 2014 at 3:46 PM,  wrote:
>> Yes, it is capable.
>> 
>> On 08 Oct 2014, at 15:42, Satish Patel  wrote:
>> 
>> > Hi,
>> >
>> > Just want to know does CDRTool prepaid capable of handling couple hundreds 
>> > of concurrent calls? I heard it can handle only 2/3 concurrent calls per 
>> > account?  what is the solution if we want to host big prepaid system with 
>> > thousands of users?
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>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> 
>> 
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Re: [OpenSIPS-Users] CDRTool prepaid for big installation

[Adrian Georgescu]

On 12 Oct 2014, at 09:48, Satish Patel  wrote:

> I have run sipp test and it only able to handle 30 calls and later all call 
> failed,

Can you explain what exactly failed?

> I heard from other user post, CDRTool prepaid can't handle many simultaneous 
> running calls.

You heard wrong. It cannot handle high density call attempts like calls 
generated from call centers or transit peers like SIP trunks that push lot of 
calls. The number of simultaneous calls is irrelevant. You can have thousands 
of simultaneous calls with almost no performance penalty if the traffic is 
generated by regular SIP user devices.

> In our case single account will make many simultaneous calls and we need to 
> handle them via prepaid.. 

It all depends on the meaning of many. Whenever a new call is attempted, the 
maximum remaining time of all ongoing calls of the same user must be 
recalculated so that the balance cannot be exceeded for any of them. This means 
that the more calls for the same user you have, the longer it takes to 
calculated everything over and over again.

If you have many users with a few calls each like in a residential scenario 
where a user makes one or perhaps two parallel calls, this would have little 
impact as there is little to re-calculate.

> Some one suggested don't use prepaid because of limitation and performance, 
> and suggested use Postpaid or Quota system..  is that true?  

It all depends on the traffic patterns. Concurrent or simultaneous calls is one 
thing, high density calls/per second attempts is another. There is no hard 
limitation but the number of database queries, distance to MySQL database will 
affect how many calls you can handle because as I explained before all 
concurrent calls must be rerated in real time again for each new call attempt. 
If one SIP account generates 10K parallel calls the load is infinite while if 
you have 10K users with one call each the load is almost zero.

This is why a prepaid model is not practical for high density of calls and this 
has little to do with CDRTool, any other system would face the same problem, 
the load is compounded when adding more calls for same account. A quota based 
system is more appropriate for entities that generate large amount of calls as 
nothing has to be calculated on a per call basis.

Adrian

> On Thu, Oct 9, 2014 at 3:46 PM,  wrote:
> Yes, it is capable.
> 
> On 08 Oct 2014, at 15:42, Satish Patel  wrote:
> 
> > Hi,
> >
> > Just want to know does CDRTool prepaid capable of handling couple hundreds 
> > of concurrent calls? I heard it can handle only 2/3 concurrent calls per 
> > account?  what is the solution if we want to host big prepaid system with 
> > thousands of users?
> > ___
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> 
> 
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Re: [OpenSIPS-Users] help: nathelper/rtpproxy/ice/ice-mismatch

[Adrian Georgescu]

Try MediaProxy, it handles ICE negotiation properly.

Adrian

On 11 Oct 2014, at 10:26, george wu  wrote:

> When I use nathelper/rtpproxy, ice does not work well with it.
> nathelper will rewrite the sdp media part which my ice client is not happy.
> Then it will reply a=ice-mismatch. finally it will use rtpproxy to relay the 
> media.
> 
> the invite:
> m=audio 36580 RTP/AVP 124 120 111 110 0 8 101
> 
> a=candidate:1 1 UDP 2130706431 192.168.1.3 7078 typ host
> a=candidate:1 2 UDP 2130706430 192.168.1.3 7079 typ host
> 
> Detail is below:
> 
> 
> 1) I have set up the nathelper/rtpproxy as below:
>   NAT modules
> loadmodule "nathelper.so"
> modparam("nathelper", "natping_interval", 10)
> modparam("nathelper", "ping_nated_only", 1)
> modparam("nathelper", "received_avp", "$avp(received_nh)")
> 
> loadmodule "rtpproxy.so"
> modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:12221") # CUSTOMIZE ME
> 
> loadmodule "stun.so"
> modparam("stun", "primary_ip", "192.168.1.3")
> modparam("stun","alternate_ip","192.168.122.1")
> 
> 2) my client is linphone with ice set up.
> 3) when it make a call with ice, the sdp media get relayed:
> INVITE sip:test2@192.168.1.3:5080 SIP/2.0
> Record-Route: 
> Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bKa00d.25386e01.0
> Via: SIP/2.0/UDP 
> 192.168.1.3:5070;received=192.168.1.3;branch=z9hG4bK.bo~6nN-3E;rport=5070
> From: ;tag=0tEh0Q~ly
> To: sip:test2@192.168.1.3
> CSeq: 20 INVITE
> Call-ID: XvJ7qOUjnp
> Max-Forwards: 69
> Supported: replaces, outbound
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, 
> INFO
> Content-Type: application/sdp
> Content-Length: 547
> Contact: 
> ;+sip.instance=""
> User-Agent: linphone/3.7.0 (belle-sip/1.3.0)
> 
> v=0
> o=test1 1936 2136 IN IP4 192.168.1.3
> s=Talk
> c=IN IP4 192.168.1.3
> t=0 0
> a=ice-pwd:741608ce2b68ba853500cdf3
> a=ice-ufrag:0d47513b
> m=audio 36580 RTP/AVP 124 120 111 110 0 8 101
> a=rtpmap:124 opus/48000
> a=fmtp:124 useinbandfec=1; usedtx=1
> a=rtpmap:120 SILK/16000
> a=rtpmap:111 speex/16000
> a=fmtp:111 vbr=on
> a=rtpmap:110 speex/8000
> a=fmtp:110 vbr=on
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=candidate:1 1 UDP 2130706431 192.168.1.3 7078 typ host
> a=candidate:1 2 UDP 2130706430 192.168.1.3 7079 typ host
> a=nortpproxy:yes
> //
> SIP/2.0 200 Ok
> Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bKa00d.25386e01.0
> Via: SIP/2.0/UDP 
> 192.168.1.3:5070;received=192.168.1.3;branch=z9hG4bK.bo~6nN-3E;rport=5070
> From: ;tag=0tEh0Q~ly
> To: ;tag=OsFF6CF
> Call-ID: XvJ7qOUjnp
> CSeq: 20 INVITE
> User-Agent: linphone/3.7.0 (belle-sip/1.3.0)
> Supported: replaces, outbound
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, 
> INFO
> Contact: 
> ;+sip.instance=""
> Content-Type: application/sdp
> Content-Length: 428
> Record-route: 
> 
> v=0
> o=test2 2088 1279 IN IP4 192.168.1.3
> s=Talk
> c=IN IP4 192.168.1.3
> t=0 0
> a=ice-pwd:52c18fd43896c0d573decfcd
> a=ice-ufrag:5d79a96c
> m=audio 7088 RTP/AVP 124 120 111 110 0 8 101
> a=rtpmap:124 opus/48000
> a=fmtp:124 useinbandfec=1; usedtx=1
> a=rtpmap:120 SILK/16000
> a=rtpmap:111 speex/16000
> a=fmtp:111 vbr=on
> a=rtpmap:110 speex/8000
> a=fmtp:110 vbr=on
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ice-mismatch
> 
> 
> 
> 
> 
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Re: [OpenSIPS-Users] CDRtool

[Adrian Georgescu]

Rating in real time involves many database and other operations depending on 
the complexity of the rating tables. Pushing 60 calls per second in a 
residential scenario will yield a lot of simultaneous calls. You can handle the 
traffic of many residential subscribers at 60 cps. I am personally surprised 
you were able to do this on one machine. 

You are trying to use the application in a high density trunking scenario and 
is no wonder you hit a limit, the application was not designed for being used 
in such scenario. The idea was to be able to handle 2/3 calls for discrete SIP 
devices belongs to one SIP account, where each SIP account corresponds to a SIP 
device like a cable modem used by residential subscribers. You are pushing it 
to do things beyond its design envelope. 

You should better use the quota system with postpaid rather than real time 
prepaid for what you describe as it makes little sense to have prepaid balance 
calculated in real time for high density trunks that generate lots of traffic 
and associated costs.

Adrian

On 30 Sep 2014, at 09:57, Pavel Eremin  wrote:

> Thanks, i read all of docs and have question about performance. (I Did 
> describe it in another email to community)
> 
> 30.09.2014 4:56 пользователь "Adrian Georgescu"  
> написал:
> If you want to dive this deep into CDRTool, it would be a good start to read 
> the documentation, that is all *.txt files in docs/ folder.
> 
> --
> Adrian
> 
> 
> 
> 
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Re: [OpenSIPS-Users] CDRtool

[Adrian Georgescu]

If you want to dive this deep into CDRTool, it would be a good start to read 
the documentation, that is all *.txt files in docs/ folder.

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Re: [OpenSIPS-Users] CDRTool E.164 number

[Adrian Georgescu]

On 22 Aug 2014, at 17:24, Satish Patel  wrote:

> CDRTool driving me crazy... I have setup CDR with Call_control module and now 
> problem is international number.. 
> 
> My first question how do i configure CDRTool for global word.. I don't just 
> want to use Europe E.164 or US E.164 numbers... I want each and every country 
> number. 

Those classes just define how people from Europe and US use to dial to "every 
country number". In Europe people use to dial 00, then country code, then the 
number.  In US is a bit different. 

> People saying configure cdr_generic.php file to set E164_class but why i set 
> that class, I want world wide numbers, anybody can dial anywhere.. 

People told you the right thing, you can dial any number in any country. You 
just miss understood the meaning of it.

> I want to dial US number and i have configured destination_id = 1 and set 
> same destination in rate table.. I can also test that using telnet to CDR 
> port like following

If you use European style dialing style, that is the Europe E.164 class, you 
must use 001 to match US numbers. 

> 
> ShowPrice From=sip:3...@sip.com To=sip:11646xxx...@sip.com 
> Gateway=174.236.4.168 Duration=59
> 0.0500
> Increment: 6 s
>  Min duration: 6 s
> Duration: 60 s
>  App: audio
>  Destination: 1
> Customer: subscriber=3...@sip.com
>  Connect: 0.
>StartTime: 2014-08-22 22:07:46
> --
> Span: 1
> Duration: 60 s
>ProfileId: sip_profile / weekday
>   RateId: sip_profile / 0-24h
> Rate: 0.0500 / 60 s
>Price: 0.0500
> 
> 
> but when i dial that number i am getting following error from call control, 
> why my DestId=None ?? 
> 
> cdrtool[17524]: MaxSessionTime Duration=36000 
> CallId=miN0QR-M_WeFgBrC94RpEw..-S From=sip:3...@sip.com 
> Gateway=174.xxx.xxx.xxx To=sip:11646...@sip.com
> cdrtool[17524]: MaxSessionTime=unlimited Type=prepaid 
> CallId=miN0QR-M_WeFgBrC94RpEw..-S BillingParty=3...@sip.com DestId=None
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Re: [OpenSIPS-Users] create mediaproxy certificates

[Adrian Georgescu]

Just google SSL tutorial or how to and you can find tons of examples

http://www.openssl.org/docs/HOWTO/certificates.txt

Adrian

On 15 Jul 2014, at 19:51, Edwin  wrote:

> Hi Saúl,
> 
> In fact I have installed opensips and mediaproxy-dispatcher on Debian and
> the mediaproxy-relay on two other Debian servers. But tinyca2 had to run on
> a server with a Gui (like KDE or Gnome), which I (rather) don't run.
> 
> I rather just use the cli, I think we only need four or five commands (but
> you just need to know which ones :) )
> 
> Regard,
> 
> Edwin
> 
> 
> 
> --
> View this message in context: 
> http://opensips-open-sip-server.1449251.n2.nabble.com/create-mediaproxy-certificates-tp7592419p7592425.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
> 
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Re: [OpenSIPS-Users] MediaProxy scalability FAQ updated

[Adrian Georgescu]

Some people reportedly saw at least two calls, but they had a different sense 
of humor.

--
Adrian

> On 04 Mar 2014, at 14:27, david  wrote:
> 
> Mediaproxy can handle at least one simultaneous call, regardless of the 
> hardware resources available providing no other program competes with the 
> same resources on that machine. Bigger scalability can be achieved by adding 
> more hardware.
> 
> ???
> 
> Mediaproxy can handle at least one simultaneous call?
> 
>> On 3/4/14 11:15 AM, a...@ag-projects.com wrote:
>> http://mediaproxy.ag-projects.com/projects/mediaproxy/wiki/Scalability
>> 
>> Adrian
>> 
>> 
>> 
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Re: [OpenSIPS-Users] Truncating ACC and CDRs

[Adrian Georgescu]

Any operation that writes to your MySQL database for any great length of time 
like trimming your table will block anything else accesing the same database. 
You should auto rotate tables and use non transactional table storage to be 
able to trim older tables without impacting the active ones.

--
Adrian

> On 21 Feb 2014, at 16:49, David Crow  wrote:
> 
> I’ve got an opensips proxy that I just use for PSTN routing, no billing or 
> anything is done from this system.
> I only use the cdrs to identify when provider a call went to or from. 
>  
> Both the ACC and CDRs table have just been growing and I haven’t done 
> anything with them and now the acc table has gotten to over 4.6 GB and it’s 
> almost impossible to search through.  I have stopped generating the CDRs for 
> now.  I would like to just purge these tables and start over but I wasn’t 
> sure if I would break anything if I run a truncate on the acc table.  Does 
> anything read from the ACC table other than the CDR generation script?  Would 
> it be ok to run a TRUNCATE on this table? 
>  
> Thanks,
>  
> David Crow | Senior Systems Architect
> 1301 Gervais Street, Suite 1800 | Columbia, SC 29201
> (d) 803.978.2727 | (f) 803.733.5888
> david.c...@vc3.com| www.VC3.com
> Follow us: 
>  
>  
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Re: [OpenSIPS-Users] CDRTool - Rating Origination and Termination differently

[Adrian Georgescu]

I think it would be a good idea.

Adrian

On 22 Jan 2014, at 16:58, David Lee (digium)  wrote:

> Duane Larson wrote:
>> I have been playing with CDRTool for a while but I am not sure if it is
>> possible to rate Origination (Inbound) calls differently than Termination
>> (Outbound) calls from my SIP Provider.  For Origination I pay 0.0035 and
>> for Termination I pay 0.005.  Keep in mind these costs are for destination
>> "1”.
> 
> I've recently run into nearly the same situation.
> 
> I tried overloading the Sip-Application-Type field, but it's limited
> by the supportedApplicationTypes array in cdr_generic.php. There also
> seems to be hard coded logic for the different application types.
> 
> I think I will patch CDRTool so that you can have subtypes of
> application types (audio.inbound, audio.outbound, etc.). This would
> allow the billing rates to be a bit more specific for these
> situations.
> 
> Thoughts?
> -- 
> David M. Lee
> Digium, Inc. | Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at:  www.digium.com  & www.asterisk.org
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Re: [OpenSIPS-Users] IM & Presence

[Adrian Georgescu]

You can start a Chat session to any SIP address, the addresses do not have to 
be in any list, it just works.

Adrian
 

On 10 Jan 2014, at 13:04, Chandra Prakash  
wrote:

> Hi,
>  
> Is it possible if we can chat with all the registered or online users without 
> adding them in buddies or contact list ?
>  
> Thanks
> Chander
>  
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Re: [OpenSIPS-Users] Groupchat functionlity with Opensips 1.9.1

[Adrian Georgescu]

There is no GUI and there is nothing to configure, it just works from the 
moment you install it.

Adrian

On Nov 25, 2013, at 5:54 PM, troxlinux  wrote:

> Hi , sylkserver.com have a GUI ?
> 
> 
> 2013/11/25 Saúl Ibarra Corretgé 
> 
> On Nov 25, 2013, at 6:53 AM, Rajesh Babu  wrote:
> 
> > Hi All,
> >
> >I have a requirement to support Groupchat on top of SIMPLE. Can anyone 
> > throw some light on please?
> >
> 
> Hi,
> 
> Groupchat in SIMPLE is specified on draft-simple-chat 
> (http://tools.ietf.org/html/draft-ietf-simple-chat-18), you can use 
> SylkServer, an Open Source implementation of it (among other things). 
> http://sylkserver.com/
> 
> 
> Regards,
> 
> --
> Saúl Ibarra Corretgé
> AG Projects
> 
> 
> 
> 
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> 
> 
> -- 
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> 
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Re: [OpenSIPS-Users] OpenSips 1.9.1 and CDRTool

[Adrian Georgescu]

These work for me with 1.9

# global acc parameters
modparam("acc", "failed_transaction_flag", 1)
modparam("acc", "report_cancels", 0)
modparam("acc", "early_media",0)

modparam("acc", "log_level",  1)
modparam("acc", "log_flag",   1)
modparam("acc", "log_missed_flag",1)

modparam("acc", "aaa_url","radius:/etc/opensips/radius/client.conf")
modparam("acc", "aaa_flag",   1)
modparam("acc", "aaa_missed_flag",1)
modparam("acc", "aaa_extra",  "User-Name=$Au; \
   
Calling-Station-Id=$avp(calling_station); \
   Called-Station-Id=$to; \
   Sip-Translated-Request-URI=$ru; \
   Sip-RPid=$avp(asserted_identity); \
   Source-IP=$avp(source_ip); \
   Source-Port=$avp(source_port); \
   SIP-Proxy-IP=$avp(sip_proxy_ip); \
   Canonical-URI=$avp(can_uri); \
   Billing-Party=$avp(billing_party); \
   Divert-Reason=$avp(diversion_reason); \
   User-Agent=$hdr(user-agent); \
   Contact=$hdr(contact); \
   Event=$hdr(event); \
   ENUM-TLD=$avp(enum_tld); \
   From-Header=$hdr(from); \
   
SIP-Application-Type=$avp(sip_application_type)")


On Jul 17, 2013, at 12:31 PM, Miguel J. López Valverde  
wrote:

> Hello
> 
> I'm trying to set up a test model with OpenSIPS 1.9.1 and CDRTool, but the 
> information available on the link 
> http://cdrtool.ag-projects.com/projects/cdrtool/wiki/Installation_Guide is 
> only valid up to version 1.6 OpenSIPS, generating errors with the use of acc 
> module parameters.
> 
> I searched information in different forums and sources, but can not find 
> anything suitable.
> 
> Can you recommend me some valid information source for working with these 
> versions?, (Links, courses, training low down payment, etc).
> 
> Thank you very much, best regards.
> -- 
> 
> 
> 
> -
> Sus datos de carácter personal (nombre, apellidos, dirección postal y de 
> correo electrónico, etc.) son tratados para la gestión de su relación con la 
> Entidad, así como para el envío de información sobre nuestra actividad y la 
> de terceros relacionadas con la actividad de Consulting Smartic Solutions, 
> S.L., CIF: B85130037, C/Pº de la Castellana, 135, 7ª planta, 28046 Madrid. 
> Usted puede ejercer sus derechos de acceso, rectificación, cancelación y 
> oposición dirigiéndose por escrito, con copia de un documento que acredite su 
> identidad, a la dirección info (arroba) smartic.es.
> Este mensaje puede contener información confidencial. Si usted no es su 
> destinatario, no debe leerlo, copiarlo, distribuirlo, ni hacer uso de la 
> información que contiene. En este caso, por favor, llámenos o comuníquenoslo 
> por escrito y borre este mensaje de su sistema.
> - 
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[OpenSIPS-Users] OpenSIPS workshop well attended at ElastixWorld in Mexico City (about 150 people)

[Adrian Georgescu]

Some pics:

https://pbs.twimg.com/media/BWoMHObCAAIn9Dy.jpg

http://pic.twitter.com/wJXn2cmro9

Regards,
Adrian and Saul



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[OpenSIPS-Users] OpenSIPS at Elastix world

[Adrian Georgescu]

Hello,

At Elastix World event in Mexico City, myself and Saul will provide a workshop 
about running OpenSIPS based SIP platforms.

http://www.elastixworld.com/2013/index.php/en/about-ew13/workshops.html#AGPRO

Main topic will be deploying SIP application beyond VoIP, namely chat, 
presence, file transfer and screen sharing.

Regards,
Adrian



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Re: [OpenSIPS-Users] file transfer in opensips

[Adrian Georgescu]

As far as I know, last time I checked IMS droid does not support either TLS 
transport or MSRP relay extension.

Regards,
Adrian

On Aug 8, 2013, at 8:58 AM, Nandini madhu wrote:

> Dear All,
> Greetings,
> 
> 
> have installed msrp relay in my system.by following the link.
> iam using ubuntu 10.04
> http://msrprelay.org/projects/msrprelay/wiki/InstallationGuide
> 
> The client iam using is IMS droid.
> i cannot transfer the file.
> 
> is there any configuration to be done in msrp relay.
> 
> kindly please help me.
> 
> 
> 
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Re: [OpenSIPS-Users] OpenSips 1.9.1 and CDRTool

[Adrian Georgescu]

CDRTool works fine with 1.9 as far as know, I use it. Can you paste the errors 
you have?

Adrian

On Jul 17, 2013, at 5:31 PM, Miguel J. López Valverde  
wrote:

> Hello
> 
> I'm trying to set up a test model with OpenSIPS 1.9.1 and CDRTool, but the 
> information available on the link 
> http://cdrtool.ag-projects.com/projects/cdrtool/wiki/Installation_Guide is 
> only valid up to version 1.6 OpenSIPS, generating errors with the use of acc 
> module parameters.
> 
> I searched information in different forums and sources, but can not find 
> anything suitable.
> 
> Can you recommend me some valid information source for working with these 
> versions?, (Links, courses, training low down payment, etc).
> 
> Thank you very much, best regards.
> -- 
> 
> 
> 
> -
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> correo electrónico, etc.) son tratados para la gestión de su relación con la 
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> de terceros relacionadas con la actividad de Consulting Smartic Solutions, 
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Re: [OpenSIPS-Users] Call Generator

[Adrian Georgescu]

There are such tools but it depends for what purpose. Do you want to test heavy 
load or just call flows?

Adrian

On Jun 18, 2013, at 7:03 PM, Maciej Bylica  wrote:

> Hello,
> 
> I am looking for call generator that is capable of:
> - generating and in the same time pick up the call (the call will traverse 
> infrastructure under testing and get back to generator)
> - generating SIP + RTP calls. There must be many .wav or mp3 files possible 
> to be used
> - heaving random call duration
> - heaving a possibility to set Called and Called numbers random in specified 
> ranges (like 1122233[0-9]{4} for instance).
> 
> Have you tested any call generator that has aforementioned functionality 
> implemented?
> I know that Opensips could be used for this purpose, but i am looking for the 
> ready-to-run product.
> 
> Thanks,
> Mac.
> 
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Re: [OpenSIPS-Users] Call Generator

[Adrian Georgescu]

See sipclient package, it contains this tool that does pretty much all you are 
looking for:

http://sipsimpleclient.org/projects/sipsimpleclient/wiki/Sip_audio_session

Adrian

On Jun 18, 2013, at 7:03 PM, Maciej Bylica  wrote:

> Hello,
> 
> I am looking for call generator that is capable of:
> - generating and in the same time pick up the call (the call will traverse 
> infrastructure under testing and get back to generator)
> - generating SIP + RTP calls. There must be many .wav or mp3 files possible 
> to be used
> - heaving random call duration
> - heaving a possibility to set Called and Called numbers random in specified 
> ranges (like 1122233[0-9]{4} for instance).
> 
> Have you tested any call generator that has aforementioned functionality 
> implemented?
> I know that Opensips could be used for this purpose, but i am looking for the 
> ready-to-run product.
> 
> Thanks,
> Mac.
> 
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Re: [OpenSIPS-Users] cdrtool

[Adrian Georgescu]

Please make a patch, we will analyze it and consider it for adding it to the 
main trunk.

Adrian

On Jun 15, 2013, at 12:59 AM, dotnetdub wrote:

> Hi Adrian,
> 
> Would you integrate this if we developed it?
> 
> Best
> Brian
> 
> 
> On 28 May 2013 12:44, Adrian Georgescu  wrote:
> It is not possible.
> 
> Adrian
> 
> On May 28, 2013, at 1:00 PM, dotnetdub wrote:
> 
> > Hi Guys,
> >
> > Is it possible to do quota by domain rather than user?
> >
> > Thanks
> > Brian
> > ___
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> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 
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Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid

[Adrian Georgescu]

You need to add pricing info in the other billing tables too.

Adrian

On Jun 11, 2013, at 3:24 PM, Leonardo Uzcudun wrote:

> Thanks again Adrian.
> You mean to insert in destinations (CDRtool web portal -> Rating -> 
> Destinations) the use...@example.com in the destination field?
> 
> Da: Adrian Georgescu 
> A: OpenSIPS users mailling list  
> Inviato: Martedì 11 Giugno 2013 14:32
> Oggetto: Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
> 
> Never tried it myself for prepaid but you can setup SIP URIs as valid 
> destinations too and put a price on them. 
> 
> Adrian
> 
> On Jun 11, 2013, at 12:29 PM, Leonardo Uzcudun wrote:
> 
>> I guess this confirm why i can't use prepaid for calls between sip users..., 
>> do you confirm it?
>> Thanks.
>> 
>> Da: Adrian Georgescu 
>> A: OpenSIPS users mailling list  
>> Inviato: Martedì 11 Giugno 2013 11:57
>> Oggetto: Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
>> 
>> The prepaid engine was designed for PSTN destinations using phone numbers 
>> only. 
>> 
>> Adrian
>> 
>> On 11 Jun 2013, at 11:49, Leonardo Uzcudun  wrote:
>> 
>>> Hello Adrian:
>>> 
>>> From the following site 
>>> (http://callcontrol.ag-projects.com/issues/1769#change-8449) i've found 
>>> that you say "All internals of CDRTool rely on the fact that there is a 
>>> normalized URI in the form of 00 + Country Code + Number."
>>> So, my question or doubt is:
>>> If there are 2 sip users within the same domain like use...@example.com and 
>>> use...@example.com, can i implement prepaid billing between them or the 
>>> prepaid will only work if they call to 00 + Country Code + Number sip 
>>> accounts?
>>> Thanks,
>>> 
>>> Leo
>>> Da: Adrian Georgescu 
>>> A: OpenSIPS users mailling list  
>>> Inviato: Lunedì 10 Giugno 2013 21:39
>>> Oggetto: Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
>>> 
>>> There is Rating.txt document that contain sample rates files and examples 
>>> of prepaid queries in the console that shows you how it works. 
>>> 
>>> Adrian
>>> 
>>> 
>>> On Jun 10, 2013, at 9:30 PM, Leonardo Uzcudun wrote:
>>> 
>>>> Hello All:
>>>> 
>>>> After doing more tests i guess the problem is the missing DestId
>>>> cdrtool log:
>>>> Jun  8 02:51:12 sip cdrtool[13359]: MaxSessionTime=unlimited Type=prepaid 
>>>> CallId=1226626922 BillingParty=user01@mydomain DestId=None
>>>> 
>>>> and considering the rating.txt doc point 3. Determination of the costs -> 
>>>> No rate will be returned if no 'destination id' is found.
>>>> 
>>>> So, now my question is: If the Destination (cdrtool web portal -> Rating 
>>>> -> Destinations) must be a number (like 31 for the Netherlands), how will 
>>>> i match destination to mydomain?
>>>> 
>>>> Thanks,
>>>> 
>>>> Leo.
>>>> Da: leo 
>>>> A: users@lists.opensips.org 
>>>> Inviato: Venerdì 7 Giugno 2013 9:38
>>>> Oggetto: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
>>>> 
>>>> Hello:
>>>> 
>>>> I need some help to understand the CDRTool - Callcontrol - Prepaid, all the
>>>> efforts i could do reading, searching and testing to try to understand how
>>>> it works were unsuccessful.
>>>> 
>>>> My set-up would be very simple, I would like to give credit (in seconds) to
>>>> the users and after the call is ended the call duration time should be
>>>> subtracted to its balance.
>>>> 
>>>> My first concern is how do i map a user in the opensips.subscriber
>>>> (DB.table) to the prepaid billing?
>>>> The second one would be, how would i configure the rating/quota/prepaid to
>>>> have my set-up?
>>>> 
>>>> Thanks for your help.
>>>> 
>>>> Leo.
>>>> 
>>>> 
>>>> 
>>>> --
>>>> View this message in context: 
>>>> http://opensips-open-sip-server.1449251.n2.nabble.com/CDRTool-Callcontrol-Prepaid-tp7586750.html
>>>> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>>>> 
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Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid

[Adrian Georgescu]

Never tried it myself for prepaid but you can setup SIP URIs as valid 
destinations too and put a price on them. 

Adrian

On Jun 11, 2013, at 12:29 PM, Leonardo Uzcudun wrote:

> I guess this confirm why i can't use prepaid for calls between sip users..., 
> do you confirm it?
> Thanks.
> 
> Da: Adrian Georgescu 
> A: OpenSIPS users mailling list  
> Inviato: Martedì 11 Giugno 2013 11:57
> Oggetto: Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
> 
> The prepaid engine was designed for PSTN destinations using phone numbers 
> only. 
> 
> Adrian
> 
> On 11 Jun 2013, at 11:49, Leonardo Uzcudun  wrote:
> 
>> Hello Adrian:
>> 
>> From the following site 
>> (http://callcontrol.ag-projects.com/issues/1769#change-8449) i've found that 
>> you say "All internals of CDRTool rely on the fact that there is a 
>> normalized URI in the form of 00 + Country Code + Number."
>> So, my question or doubt is:
>> If there are 2 sip users within the same domain like use...@example.com and 
>> use...@example.com, can i implement prepaid billing between them or the 
>> prepaid will only work if they call to 00 + Country Code + Number sip 
>> accounts?
>> Thanks,
>> 
>> Leo
>> Da: Adrian Georgescu 
>> A: OpenSIPS users mailling list  
>> Inviato: Lunedì 10 Giugno 2013 21:39
>> Oggetto: Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
>> 
>> There is Rating.txt document that contain sample rates files and examples of 
>> prepaid queries in the console that shows you how it works. 
>> 
>> Adrian
>> 
>> 
>> On Jun 10, 2013, at 9:30 PM, Leonardo Uzcudun wrote:
>> 
>>> Hello All:
>>> 
>>> After doing more tests i guess the problem is the missing DestId
>>> cdrtool log:
>>> Jun  8 02:51:12 sip cdrtool[13359]: MaxSessionTime=unlimited Type=prepaid 
>>> CallId=1226626922 BillingParty=user01@mydomain DestId=None
>>> 
>>> and considering the rating.txt doc point 3. Determination of the costs -> 
>>> No rate will be returned if no 'destination id' is found.
>>> 
>>> So, now my question is: If the Destination (cdrtool web portal -> Rating -> 
>>> Destinations) must be a number (like 31 for the Netherlands), how will i 
>>> match destination to mydomain?
>>> 
>>> Thanks,
>>> 
>>> Leo.
>>> Da: leo 
>>> A: users@lists.opensips.org 
>>> Inviato: Venerdì 7 Giugno 2013 9:38
>>> Oggetto: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
>>> 
>>> Hello:
>>> 
>>> I need some help to understand the CDRTool - Callcontrol - Prepaid, all the
>>> efforts i could do reading, searching and testing to try to understand how
>>> it works were unsuccessful.
>>> 
>>> My set-up would be very simple, I would like to give credit (in seconds) to
>>> the users and after the call is ended the call duration time should be
>>> subtracted to its balance.
>>> 
>>> My first concern is how do i map a user in the opensips.subscriber
>>> (DB.table) to the prepaid billing?
>>> The second one would be, how would i configure the rating/quota/prepaid to
>>> have my set-up?
>>> 
>>> Thanks for your help.
>>> 
>>> Leo.
>>> 
>>> 
>>> 
>>> --
>>> View this message in context: 
>>> http://opensips-open-sip-server.1449251.n2.nabble.com/CDRTool-Callcontrol-Prepaid-tp7586750.html
>>> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>>> 
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>> 
>>> 
>>> ___
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>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> 
>> 
>> ___
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> 
>> 
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 
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Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid

[Adrian Georgescu]

The prepaid engine was designed for PSTN destinations using phone numbers only. 

Adrian

On 11 Jun 2013, at 11:49, Leonardo Uzcudun  wrote:

> Hello Adrian:
> 
> From the following site 
> (http://callcontrol.ag-projects.com/issues/1769#change-8449) i've found that 
> you say "All internals of CDRTool rely on the fact that there is a normalized 
> URI in the form of 00 + Country Code + Number."
> So, my question or doubt is:
> If there are 2 sip users within the same domain like use...@example.com and 
> use...@example.com, can i implement prepaid billing between them or the 
> prepaid will only work if they call to 00 + Country Code + Number sip 
> accounts?
> Thanks,
> 
> Leo
> Da: Adrian Georgescu 
> A: OpenSIPS users mailling list  
> Inviato: Lunedì 10 Giugno 2013 21:39
> Oggetto: Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
> 
> There is Rating.txt document that contain sample rates files and examples of 
> prepaid queries in the console that shows you how it works. 
> 
> Adrian
> 
> 
> On Jun 10, 2013, at 9:30 PM, Leonardo Uzcudun wrote:
> 
>> Hello All:
>> 
>> After doing more tests i guess the problem is the missing DestId
>> cdrtool log:
>> Jun  8 02:51:12 sip cdrtool[13359]: MaxSessionTime=unlimited Type=prepaid 
>> CallId=1226626922 BillingParty=user01@mydomain DestId=None
>> 
>> and considering the rating.txt doc point 3. Determination of the costs -> No 
>> rate will be returned if no 'destination id' is found.
>> 
>> So, now my question is: If the Destination (cdrtool web portal -> Rating -> 
>> Destinations) must be a number (like 31 for the Netherlands), how will i 
>> match destination to mydomain?
>> 
>> Thanks,
>> 
>> Leo.
>> Da: leo 
>> A: users@lists.opensips.org 
>> Inviato: Venerdì 7 Giugno 2013 9:38
>> Oggetto: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
>> 
>> Hello:
>> 
>> I need some help to understand the CDRTool - Callcontrol - Prepaid, all the
>> efforts i could do reading, searching and testing to try to understand how
>> it works were unsuccessful.
>> 
>> My set-up would be very simple, I would like to give credit (in seconds) to
>> the users and after the call is ended the call duration time should be
>> subtracted to its balance.
>> 
>> My first concern is how do i map a user in the opensips.subscriber
>> (DB.table) to the prepaid billing?
>> The second one would be, how would i configure the rating/quota/prepaid to
>> have my set-up?
>> 
>> Thanks for your help.
>> 
>> Leo.
>> 
>> 
>> 
>> --
>> View this message in context: 
>> http://opensips-open-sip-server.1449251.n2.nabble.com/CDRTool-Callcontrol-Prepaid-tp7586750.html
>> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>> 
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> 
>> 
>> ___
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 
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Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid

[Adrian Georgescu]

There is Rating.txt document that contain sample rates files and examples of 
prepaid queries in the console that shows you how it works. 

Adrian


On Jun 10, 2013, at 9:30 PM, Leonardo Uzcudun wrote:

> Hello All:
> 
> After doing more tests i guess the problem is the missing DestId
> cdrtool log:
> Jun  8 02:51:12 sip cdrtool[13359]: MaxSessionTime=unlimited Type=prepaid 
> CallId=1226626922 BillingParty=user01@mydomain DestId=None
> 
> and considering the rating.txt doc point 3. Determination of the costs -> No 
> rate will be returned if no 'destination id' is found.
> 
> So, now my question is: If the Destination (cdrtool web portal -> Rating -> 
> Destinations) must be a number (like 31 for the Netherlands), how will i 
> match destination to mydomain?
> 
> Thanks,
> 
> Leo.
> Da: leo 
> A: users@lists.opensips.org 
> Inviato: Venerdì 7 Giugno 2013 9:38
> Oggetto: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
> 
> Hello:
> 
> I need some help to understand the CDRTool - Callcontrol - Prepaid, all the
> efforts i could do reading, searching and testing to try to understand how
> it works were unsuccessful.
> 
> My set-up would be very simple, I would like to give credit (in seconds) to
> the users and after the call is ended the call duration time should be
> subtracted to its balance.
> 
> My first concern is how do i map a user in the opensips.subscriber
> (DB.table) to the prepaid billing?
> The second one would be, how would i configure the rating/quota/prepaid to
> have my set-up?
> 
> Thanks for your help.
> 
> Leo.
> 
> 
> 
> --
> View this message in context: 
> http://opensips-open-sip-server.1449251.n2.nabble.com/CDRTool-Callcontrol-Prepaid-tp7586750.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
> 
> ___
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 
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Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid

[Adrian Georgescu]

You need to add the prepaid user and its balance to cdrtool prepaid table too.

Adrian

On Jun 10, 2013, at 2:17 PM, Leonardo Uzcudun wrote:

> Thanks Adrian for replying me but how do I enable it? The only note i've 
> found in the cdrtool and callcontrol was to add the quote column to the 
> opensips.subscriber (that i've already added it). Should i add another column?
> 
> 
> Da: Adrian Georgescu 
> A: Leonardo Uzcudun ; OpenSIPS users mailling list 
>  
> Inviato: Lunedì 10 Giugno 2013 13:45
> Oggetto: Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
> 
> You must mark the user as prepaid in OpenSIPS subscribers table.
> 
> Adrian
> 
> On Jun 8, 2013, at 7:26 PM, Leonardo Uzcudun wrote:
> 
>> I guess, i've got some progress...but i can't understand why is not working.
>> Form the cdrtool logs i've:
>> 
>> Jun  8 02:51:12 sip cdrtool[13359]: MaxSessionTime Duration=36000 
>> CallId=1226626922 From=sip:user01@mydomain Gateway=X.X.X.X 
>> To=sip:user02@mydomain
>> Jun  8 02:51:12 sip cdrtool[13359]: MaxSessionTime=unlimited Type=prepaid 
>> CallId=1226626922 BillingParty=user01@mydomain DestId=None
>> now cdrtool says: Type=prepaid
>> 
>> but from the callcontrol:
>> Jun  8 02:51:12 sip call-control[29789]: Call id 1226626922 of 
>> user01@mydomain to sip:user02@mydomain is postpaid not limited
>> Why callcontrol has identified it as "postpaid not limited"?
>> 
>> My configuration is for a sigle domain, no PSTN.
>> Thanks for your help if someone could
>> 
>> 
>> Da: leo 
>> A: users@lists.opensips.org 
>> Inviato: Venerdì 7 Giugno 2013 9:38
>> Oggetto: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
>> 
>> Hello:
>> 
>> I need some help to understand the CDRTool - Callcontrol - Prepaid, all the
>> efforts i could do reading, searching and testing to try to understand how
>> it works were unsuccessful.
>> 
>> My set-up would be very simple, I would like to give credit (in seconds) to
>> the users and after the call is ended the call duration time should be
>> subtracted to its balance.
>> 
>> My first concern is how do i map a user in the opensips.subscriber
>> (DB.table) to the prepaid billing?
>> The second one would be, how would i configure the rating/quota/prepaid to
>> have my set-up?
>> 
>> Thanks for your help.
>> 
>> Leo.
>> 
>> 
>> 
>> --
>> View this message in context: 
>> http://opensips-open-sip-server.1449251.n2.nabble.com/CDRTool-Callcontrol-Prepaid-tp7586750.html
>> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>> 
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> 
>> 
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 
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Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid

[Adrian Georgescu]

You must mark the user as prepaid in OpenSIPS subscribers table.

Adrian

On Jun 8, 2013, at 7:26 PM, Leonardo Uzcudun wrote:

> I guess, i've got some progress...but i can't understand why is not working.
> Form the cdrtool logs i've:
> 
> Jun  8 02:51:12 sip cdrtool[13359]: MaxSessionTime Duration=36000 
> CallId=1226626922 From=sip:user01@mydomain Gateway=X.X.X.X 
> To=sip:user02@mydomain
> Jun  8 02:51:12 sip cdrtool[13359]: MaxSessionTime=unlimited Type=prepaid 
> CallId=1226626922 BillingParty=user01@mydomain DestId=None
> now cdrtool says: Type=prepaid
> 
> but from the callcontrol:
> Jun  8 02:51:12 sip call-control[29789]: Call id 1226626922 of 
> user01@mydomain to sip:user02@mydomain is postpaid not limited
> Why callcontrol has identified it as "postpaid not limited"?
> 
> My configuration is for a sigle domain, no PSTN.
> Thanks for your help if someone could
> 
> 
> Da: leo 
> A: users@lists.opensips.org 
> Inviato: Venerdì 7 Giugno 2013 9:38
> Oggetto: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
> 
> Hello:
> 
> I need some help to understand the CDRTool - Callcontrol - Prepaid, all the
> efforts i could do reading, searching and testing to try to understand how
> it works were unsuccessful.
> 
> My set-up would be very simple, I would like to give credit (in seconds) to
> the users and after the call is ended the call duration time should be
> subtracted to its balance.
> 
> My first concern is how do i map a user in the opensips.subscriber
> (DB.table) to the prepaid billing?
> The second one would be, how would i configure the rating/quota/prepaid to
> have my set-up?
> 
> Thanks for your help.
> 
> Leo.
> 
> 
> 
> --
> View this message in context: 
> http://opensips-open-sip-server.1449251.n2.nabble.com/CDRTool-Callcontrol-Prepaid-tp7586750.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
> 
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Re: [OpenSIPS-Users] cdrtool

[Adrian Georgescu]

It is not possible.

Adrian

On May 28, 2013, at 1:00 PM, dotnetdub wrote:

> Hi Guys,
> 
> Is it possible to do quota by domain rather than user?
> 
> Thanks
> Brian
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Re: [OpenSIPS-Users] Mediaproxy wrong port, bug?

[Adrian Georgescu]

Open a issue here

http://projects.ag-projects.com/projects/mediaproxy/issues

On May 28, 2013, at 12:53 PM, Jerzy Ptak wrote:

> Hi,
> 
> I have spent some time trying to debug this issue.
> Despite the fact that depending on the individual case you can find 
> workarounds, this is a mediaproxy issue. The problem is caused by the way 
> mediproxy is processing media packets and setting up forwarding rules.
> 
> Before the forwarding rules are set up, mediaproxy waits and passes on media 
> packets between both sides of the session. When mediaproxy receives at least 
> on media packet from each side then it is setting up forwarding rules. It 
> seems to be that, because the forwarding rules setup is not atomic there is a 
> time window in which media packet that have been handled by mediaproxy will 
> be sent out by the system (kernel) from the incorrect port.
> 
> I have created a fix for this issue, which I am allowed to contribute back to 
> mediaproxy. Can somebody from mediaproxy team let me know where can I submit 
> the fix.
> 
> 
> Regards
> Jerzy Ptak
> 
> -- 
> Jerzy Ptak
> Purple Communications Europe Ltd.
> Office: +44-1275-371700
> Email: jerzy.p...@purple.us
> 
> 
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Re: [OpenSIPS-Users] OpenSIPS and TLS with wildcard certificates again

[Adrian Georgescu]

The client must load same CA chain that signed the server cert.

Adrian

On Sep 14, 2012, at 2:13 PM, Peter Lemenkov wrote:

> Hello All!
> 
> First of all - I've read a bit about TLS and certificates in OpenSIPS
> but I still don't have a clue what's wrong with this.
> 
> My problem is - although openssl can verify certificate as well as it
> can be loaded by opensips, client apps are refusing to connect.
> Namely, empathy and Jitsi.
> 
> My setup is quite simple (well, I thought so). I've got a bunch of SIP
> domains, lets,say sip0[0-9].domain.com fully resolvable via DNS (w/o
> additional DNS SRV records - just domain names). I've got wildcard SSL
> certificate from Thawte (for "*.domain.com" without quotes) and a CA
> bundle from Thawte (
> https://search.thawte.com/library/VERISIGN/ALL_OTHER/thawte%20ca/SSL_CA_Bundle.pem
> ). I appended it to the end of the system-wide certificate bundle (and
> checked with openssl). And now here is my relevant config data (I
> added "192.168.0.1 sip01.domain.com" to /etc/hosts for the sake of
> simplicity):
> 
> disable_tls = 0
> listen = tls:192.168.0.1:5051
> tls_verify_server = 0
> tls_verify_client = 0
> tls_require_client_certificate = 0
> tls_method = TLSv1
> 
> alias=sip01.domain.com:5051
> 
> tls_certificate = "./wildcard.domain.com.crt"
> tls_private_key = "./wildcard.domain.com.key"
> tls_ca_list = "./ca-bundle.crt" # system-wide CA bundle + SSL_CA_Bundle.pem
> 
> 
> All I got so far is
> 
> Sep 14 16:02:29 [14877] ERROR:core:tls_accept: New TLS connection from
> 192.168.0.2:59588 failed to accept: rejected by client
> 
> Here is a confirmation from openssl:
> 
> work ~/work/OpenSIPS (git::1.8.x-ipport): openssl verify -CAfile
> ./ca-bundle.crt ./wildcard.domain.com.crt
> ./wildcard.domain.com.crt: OK
> work ~/work/OpenSIPS (git::1.8.x-ipport):
> 
> I'm using the same certificate for https and it works quite fine in
> Firefox. What did I miss so far?
> 
> -- 
> With best regards, Peter Lemenkov.
> 
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[OpenSIPS-Users] New release for Blink Qt 0.3.1 for Linux with SIP SIMPLE Presence (RLS and XCAP)

[Adrian Georgescu]

Hello,

There is a new release for Blink Qt for Linux available, version 0.3.1. This 
version features SIMPLE presence and bug fixes.

blink (0.3.1) unstable; urgency=low

  * Added SIMPLE Presence (PUBLISH, RLS SUBSCRIBE/NOTIFY)
  * Added XCAP storage for contacts, icon and and presence policy
  * Added handing for presence, presence.winfo and xcap-diff events
  * Added call history menu items (Missed, Received, Placed)
  * Made sRTP optional by default
  * Adjusted default timeouts for registration, subscription and publication
  * Removed/diabled some GUI elements for not yet implemented features

More information at:

http://projects.ag-projects.com/news/57

Regards,
Adrian

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Re: [OpenSIPS-Users] CDRTool without Mediaproxy

[Adrian Georgescu]

To display call detail records you need to enable Radius accounting. Radius 
authentication and authorization is not needed.

Adrian

On Apr 23, 2013, at 6:01 PM, Artur Staniec wrote:

> Hi,
> 
> my configuration of opensips uses RTPProxy, but I'd also like to use CDRTool. 
> It's a proxy so it doesn't require radius as the authentication is done on 
> the PBXs and security is IP based.
> 
> So my questions are:
> 1. Is CDRTool possible to configure without these or with some dummy Radius 
> that won't really authenticate anyone?
> 
> 2. Would I be able to get any useful data from the CDRTool like the billing 
> information for example?
> 
> Thank you very much for responses.
> 
> Kind regards,
> Artur Staniec
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[OpenSIPS-Users] OpenSIPS, OpenXCAP and Blink: SIP Presence Tutorial

[Adrian Georgescu]

Here is the list with OpenSIPS tutorials updated:

http://www.opensips.org/Resources/DocsTutorials#toc25

Regards,
Adrian



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Re: [OpenSIPS-Users] [RFC] Distributed User Location

[Adrian Georgescu]

I am  running for years servers distributed in different countries part of the 
same service and nobody complained about the latency of signaling but they 
complained about media. This idea of geo-distribution is more about media path 
optimization and automatic recovery in case of connectivity failures in one 
location rather than pointing user from country X to server Y.  Any 
distribution model that is not done by a formula like Chord does, is not 
deterministic. This means that mapping must be manually provisioned and changed 
if nodes come and go. You cannot deterministically  geo-locate to the same 
server unless you hardwire this as a setting or in a database, is not 
self-learning and in case of server failure it must be changed manually. 
Signaling-wise the geo-location is less of a problem. If it takes a fews 
seconds to make a call setup is not that critical, media relay however need to 
use shortest path so. An algorithm to allocate the closest relay to the calling 
party is much more useful. However without an async core, reserving a relay 1 
second RTT-wise away is a killer for the overall CPS. Whatever you try to do 
without an async core any distribution of resources hit the performance problem 
related to the child being blocked while processing a request and the farther 
away the database or relay the worse it gets.  I think that by addressing the 
async issue, will automagically create a multitude of solutions for better 
distribution and load balancing.

Adrian

On Apr 9, 2013, at 10:17 AM, Bogdan-Andrei Iancu wrote:

> Hi,
> 
> Putting together what you said and what Adrian and Muhammad said :
> 
> Actually we may have a distributed USRLOC for 2 purposes: geo distribution 
> and load distribution - how they are approach it is a bit different.
> 
> But first let's look into the common part (for the 2 cases) : IMHO, in both 
> cases we should have the SIP part (opensips) storing the actual full 
> registration in a certain location (via USRLOC) and an upper layer, 
> distributed, to keep a mapping between users (AORs) and the location(s) they 
> are registered with. So:
>- local level - OpenSIPS doing classing registrations (a node)
>- distributed level - some other tool to keep (in a distributed fashion) 
> the mapping of AORs on the nodes
> 
> Now, here comes the difference.
> 
> If you do geo distribution, you want to keep registration as closes as 
> possible to the user. So the registration will be kept on the OpenSIPS node 
> which was contacted by the user. In this case Chord does not work (at 
> distributed level) as it has its own alg to distributed data across nodes; in 
> our case we want to control the distribution and to say what 
> data/registration stays on what node/opensips.
> 
> If you do load distribution, you want to balance all received registrations 
> across all existing nodes/opensips - in this case a Chord like approach will 
> help (as it will do the load distribution for you).
> 
> 
> As I see the solution : have the 2 layers (local and distributed) as built in 
> in OpenSIPS and additionally to be able to use different algorithms to do the 
> mapping between registrations and OpenSIPS nodes.
> 
> 
> Is the above a good approach ??
> 
> Regards,
> 
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
> 
> 
> On 04/05/2013 04:45 PM, Rudy wrote:
>> Everyone,
>> 
>>  Before we get too off topic, I think the goal should be to design
>> something truly distributed. This would be more like what Adrian
>> suggested and less like a super node / slave node scenario. The nodes
>> should be able to coordinate amongst themselves, again, similar to the
>> docs Adrian shared.
>> 
>>  One thing we will need is a consistent hashing alg. Adrian suggested
>> Chord, another that works well for us in our implementations is
>> Ketama. Either way, it needs to be able to have consistent hashing, so
>> that additions / removals of nodes do not change the location of home
>> proxy of each registered user.
>> 
>> http://en.wikipedia.org/wiki/Consistent_hashing
>> 
>> Thanks in advance,
>> --Rudy
>> Dynamic Packet
>> Toll-Free: 888.929.VOIP ( 8647 )
>> 
>> 
>> On Fri, Apr 5, 2013 at 9:39 AM, Muhammad Shahzad  
>> wrote:
>>> Well, i am not much familiar with internals of opensips, i.e. its core and
>>> modules and how they interact with each other. But as an abstract idea, i
>>> suggest that both Base Node and Super Node should be opensips modules. No
>>> change in standard registrar or usrloc modules are actually needed.
>>> 
>>> In the Super Node module, we will have,
>>> 
>>> 1. one db table to store base node addresses for monitoring the Event.
>>> 2. one db table to store data received from the Event, lets call it "Event
>>> Table".
>>> 3. one process to manage "Event Table", pretty much the same way location
>>> table is managed by usrloc module.
>>> 4. some scripting functions for opensips.cfg, to look up in "Event Table"
>>> and do SIP

Re: [OpenSIPS-Users] MediaProxy behaviour on Heavy call volume.

[Adrian Georgescu]

The best optimization would be to make OpenSIPS core work asynchronous so that 
a request won't block the processing of new requests. Until this happens is 
little you can optimize if you want distributed infra with high cps. The lowest 
hanging fruit today to achieve highest CPS is to colocate all external 
components on the same LAN with OpenSIPS.

Adrian
  
On Apr 8, 2013, at 11:32 AM, SamyGo wrote:

> Hi AG,
> I understand your point, so in this matter we should not only consider the 
> latency between dispatcher and relay but also consider the processing 
> capability of both components as high CPS might require significant resources 
> on relays as well.
> 
> Are there any particular areas which you can point out for me to optimize 
> this?
> 
> Thanks
> Sammy
> 
> On Apr 7, 2013 6:40 PM, "Adrian Georgescu"  wrote:
> As OpenSIPS core is not async, the whole chain of processing a message can 
> cause this. Practically, summing up all RTT for all your database queries, 
> DNS lookups, Radius requests, media reservations can cause this.
> 
> The farther away each component is the poorest the performance as that child 
> cannot process any new packet until is done with the previous.
> 
> Adrian
> 
> On Apr 7, 2013, at 3:27 PM, SamyGo wrote:
> 
>> Hi Sir,
>> Yes, we've optimized our opensips to have enough shared memory as well as 
>> the number of children have been increased as well but this situation is 
>> still the same.
>> 
>> What I can logically think why this is happening is as follow:
>> 
>> 1- Media-relays are in another DC,
>> 2- Dispatcher queries relays to get the port info to update the SDP with
>> 
>> The delay or slowness of relays might be something causing overall queue 
>> length to shoot up.
>> 
>> Let me know what you think,
>> 
>> --
>> Sammy
>> 
>> 
>> 
>> On Sat, Apr 6, 2013 at 2:43 AM, Muhammad Shahzad  
>> wrote:
>> Well, if you have very high CPS then typically need more opensips children 
>> processes to handle load. So you need to,
>> 
>> 1. increase no. of children.
>> 2. optimize per child and shared memory sizes.
>> 
>> BTW what does it has to do with media proxy as you are reporting SIP UDP 
>> port only gets hanged?
>> 
>> Thank you.
>> 
>> 
>> 
>> On Fri, Apr 5, 2013 at 8:42 PM, SamyGo  wrote:
>> Hello,
>> 
>> I'm working with opensips with heavy CPS, recently I added 
>> mediaproxy-dispatcher on the server and couple of relays on different 
>> servers in different Data Center. Everything worked fine until we observed 
>> that opensips SIP UDP port 5060  got huge queued packets. 
>> 
>> That obviously impacts the call processing. I want to know if this is a 
>> usual behavior? or can this be treated/tuned ?  
>> 
>> Thanks,
>> Sammy
>> 
>> 
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>> 
>> 
>> 
>> 
>> -- 
>> Mit freundlichen Grüßen
>> Muhammad Shahzad
>> ---
>> CISCO Rich Media Communication Specialist (CRMCS)
>> CISCO Certified Network Associate (CCNA)
>> Cell: +49 176 99 83 10 85
>> MSN: shari_78...@hotmail.com
>> Email: shaherya...@googlemail.com
>> 
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Re: [OpenSIPS-Users] MediaProxy behaviour on Heavy call volume.

[Adrian Georgescu]

As OpenSIPS core is not async, the whole chain of processing a message can 
cause this. Practically, summing up all RTT for all your database queries, DNS 
lookups, Radius requests, media reservations can cause this.

The farther away each component is the poorest the performance as that child 
cannot process any new packet until is done with the previous.

Adrian

On Apr 7, 2013, at 3:27 PM, SamyGo wrote:

> Hi Sir,
> Yes, we've optimized our opensips to have enough shared memory as well as the 
> number of children have been increased as well but this situation is still 
> the same.
> 
> What I can logically think why this is happening is as follow:
> 
> 1- Media-relays are in another DC,
> 2- Dispatcher queries relays to get the port info to update the SDP with
> 
> The delay or slowness of relays might be something causing overall queue 
> length to shoot up.
> 
> Let me know what you think,
> 
> --
> Sammy
> 
> 
> 
> On Sat, Apr 6, 2013 at 2:43 AM, Muhammad Shahzad  
> wrote:
> Well, if you have very high CPS then typically need more opensips children 
> processes to handle load. So you need to,
> 
> 1. increase no. of children.
> 2. optimize per child and shared memory sizes.
> 
> BTW what does it has to do with media proxy as you are reporting SIP UDP port 
> only gets hanged?
> 
> Thank you.
> 
> 
> 
> On Fri, Apr 5, 2013 at 8:42 PM, SamyGo  wrote:
> Hello,
> 
> I'm working with opensips with heavy CPS, recently I added 
> mediaproxy-dispatcher on the server and couple of relays on different servers 
> in different Data Center. Everything worked fine until we observed that 
> opensips SIP UDP port 5060  got huge queued packets. 
> 
> That obviously impacts the call processing. I want to know if this is a usual 
> behavior? or can this be treated/tuned ?  
> 
> Thanks,
> Sammy
> 
> 
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> 
> 
> 
> -- 
> Mit freundlichen Grüßen
> Muhammad Shahzad
> ---
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +49 176 99 83 10 85
> MSN: shari_78...@hotmail.com
> Email: shaherya...@googlemail.com
> 
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