Re: [SR-Users] sdp_with_transport

[Serhat Guler]

Hello,

I have also realized that weird weird behavior with sdp_with_transport
method and started using sdp_with_transport_like("RTP/SAVPF") instead.

sdp_with_transport("RTP/SAVPF") returns false for the following line: m=audio
56481 UDP/TLS/RTP/SAVPF 109 9 0 8

If anyone could clarify it, it would be perfect.

Cheers,
Serhat

On 31 January 2017 at 20:03, M. Salman  wrote:

> Hi Guys,
>
> Just wanted to clarify the following case:
>
> what should be result of sdp_with_transport("RTP/SAVPF") on line:
>
> m=audio 10231 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
>
> I am having a weird behavior for two different versions of Kamailio. I
> hope I am doing something wrong.
>
> kamailio 4.3.2 => sdp_with_transport("RTP/SAVPF") = true
> kamailio 4.4.5 => sdp_with_transport("RTP/SAVPF") = false
>
>
>
> --
> Regards
>
> M. Salman
> VoIP Professional
>
>
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Re: [SR-Users] rtpengine - Key file error

[Serhat Guler]

Hi Richard,

Thanks for your reply. I actually was trying to install the kernel modules
on a Debian 8.7.1 system. As, I couldn't find a solution, I did a clean
install without the kernel modules, and now it just runs fine (at some
point I will try to install the kernel modules again, and will keep you
updated about how it goes).

Btw, I had glib2.0.

Cheers,
Serhat

On 28 January 2017 at 18:48, Richard Fuchs <rfu...@sipwise.com> wrote:

> On 25/01/17 06:44 PM, Serhat Guler wrote:
>
> Hello,
>
> I am trying to setup my Kamailio environment on a new Debian system.
> Everything went well, except I started facing a problem with the rtpengine.
> It did install it fine and I can view the program parameters via the help
> menu; however, when I run it with this command  "rtpengine --interface
> 192.168.0.66 --listen-ng=127.0.0.1:2 -m 3 -M 35000 -L 2" or any
> other commands, it gives an error as "CRIT: Fatal error: Bad command line:
> Key file does not have group 'rtpengine'".
>
> Where is this key file it is complaining about ? I couldn't find anything
> neither in the doc nor anywhere else.
>
> The key file is the config file. It defaults to
> /etc/rtpengine/rtpengine.conf
>
> However, if no config file is explicitly specified, it ought to ignore
> errors trying to load from it. Does that file exist on your system? What's
> your glib (not glibc) version?
>
> Cheers
>
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[SR-Users] rtpengine - Key file error

[Serhat Guler]

Hello,

I am trying to setup my Kamailio environment on a new Debian system.
Everything went well, except I started facing a problem with the rtpengine.
It did install it fine and I can view the program parameters via the help
menu; however, when I run it with this command  "rtpengine --interface
192.168.0.66 --listen-ng=127.0.0.1:2 -m 3 -M 35000 -L 2" or any
other commands, it gives an error as "CRIT: Fatal error: Bad command line:
Key file does not have group 'rtpengine'".

Where is this key file it is complaining about ? I couldn't find anything
neither in the doc nor anywhere else.

Cheers,
Serhat
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[SR-Users] Kamailio IMS Secure Websocket closes connection

[Serhat Guler]

Dear all,

I am trying to make a call between 2 sipml5 client over WSS. Everything
works fine and as expected when using WS. While using WSS, the clients are
registered successfully, but when I make a call, the called party is
disconnected from the network and the caller gets "User offline" message.
There a wireshark capture here:
https://drive.google.com/file/d/0B6LMw8kmoAMYMGhrYXNUVXJySEk/view?usp=sharing

The call is made from Alice to Bob, the port Nrs: 4060: PCSCF, 5060:ICSCF,
and 6060 SCSCF. WWS is running on 4443.

The error message that is shown in Firefox console: "The connection to
wss://192.168.0.11:4443/ was interrupted while the page was loading."

Thanks in advance,
Serhat
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Re: [SR-Users] Kamailio + IMS + sipml5 call to softphone

[Serhat Guler]

Thanks for the recommendations Alberto. I'll definitely try it out and
hopefully will be able to call a softphone from webrtc client.

Cheers,
Serhat

On 1 November 2016 at 15:17, Alberto Llamas <albertollam...@gmail.com>
wrote:

> Hi Serhat,
>
> If you take a look of SDP body of your INVITES you will note that you are
> offering SRTP.
>
> What you should do from my point of view is detect when an INVITE from the
> sipml5 softphone goes to the ims-softphone or other end-point which you are
> aware doesn't support SRTP and use the RTPEngine module (combination
> of rtpengine_offer and rtpengine_answer functions).
>
> *When INVITE (sipml5 -> Kamailio -> endPoint with out SRTP) you should
> perform something like this:*
>
> rtpengine_offer("trust-address replace-origin replace-session-connection
> ICE=remove RTP/AVP");
> t_on_reply("3");
>
> *Then for replies you will need something like the route:*
>
> onreply_route[3] {
>
> if (t_check_status("183")) {
> change_reply_status("180", "Ringing");
> remove_body();
> exit;
> }
>
> if(!(status=~"[12][0-9][0-9]") || !(sdp_content()))
> return;
> rtpengine_answer("trust-address replace-origin
> replace-session-connection ICE=force");
>
> route(NATMANAGE);
> }
>
> Or other approcah if offer SRTP and when the other ends answer with a
> 488 Not Supported Here do the same.
>
> It is one way of bridging SRTP->RTP with the RTPEngine module.
>
> Regards,
>
>
> On Tue, Nov 1, 2016 at 2:48 PM, Serhat Guler <srtgu...@gmail.com> wrote:
>
>> Hi Alberto,
>>
>> Thanks for looking into this. In the expert settings of sipml5 it says
>> that disabling RTCWeb Breaker should make it compatible with softphones
>> which are not implementing SRTP, that's how I have been testing it though.
>> May I ask what attribute you looked at to get to the conclusion you have ?
>> Thanks.
>>
>> Serhat
>>
>> On 1 November 2016 at 13:39, Alberto Llamas <albertollam...@gmail.com>
>> wrote:
>>
>>> Hello Serhat,
>>>
>>> When you are using the webphone (sipml5) by WebRTC the media is secured
>>> with SRTP. So if the other end-point supports SRTP usually you don't have
>>> major issues. It is like when you communicate between two sipml5 web phones
>>> A and B.
>>>
>>> But when you are trying to communicate to the IMS softphone, be sure
>>> that the softphone supports SRTP otherwise you will need to configure a RTP
>>> Proxy like RTPEngine in kamailio module in order to "translate" between
>>> plain RTP and SRTP.
>>>
>>> This is what I see is your issue based on the pcap files.
>>>
>>> PS: You can have a setup in your kamailio config file to offer first
>>> SRTP and if the other end-point doesn't support it (when you receive a 4XX
>>> reply) then send a Re-INVITE with plain RTP.
>>>
>>> Regards,
>>>
>>> On Tue, Nov 1, 2016 at 12:45 PM, Serhat Guler <srtgu...@gmail.com>
>>> wrote:
>>>
>>>> Hi Daniel, hi Alberto,
>>>>
>>>> Thanks for your prompt replies. I have put 2 pcap files in dropbox (
>>>> https://www.dropbox.com/sh/fzclmbpniebrvx1/AAAOOv4h2ci7bJu
>>>> uJvSbs3poa?dl=0 ) . trace.mercuro.pcap is the one where the session is
>>>> set up, but there is no audio flow and trace.boghe.pcap is the one with 488
>>>> error.
>>>>
>>>> Cheers,
>>>> Serhat
>>>>
>>>> On 1 November 2016 at 12:39, Daniel-Constantin Mierla <
>>>> mico...@gmail.com> wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>> can you get the SIP INVITE content that was received by the endpoint
>>>>> returning 488? Maybe we can spot if there is something wrong in the sip
>>>>> message content or an issue in the endpoint software. Maybe it doesn't 
>>>>> like
>>>>> headers with random string instead of ip addresses (e.g., in via, contact
>>>>> ...).
>>>>>
>>>>> I am not aware of any ims softphone with webrtc capabilities.
>>>>> Cheers,
>>>>> Daniel
>>>>>
>>>>>
>>>>> On 01/11/16 12:15, Serhat Guler wrote:
>>>>>
>>>>> Hi,
>>>>>
>>>>> I have a setup as follows:

Re: [SR-Users] Kamailio + IMS + sipml5 call to softphone

[Serhat Guler]

Hi Alberto,

Thanks for looking into this. In the expert settings of sipml5 it says that
disabling RTCWeb Breaker should make it compatible with softphones which
are not implementing SRTP, that's how I have been testing it though. May I
ask what attribute you looked at to get to the conclusion you have ? Thanks.

Serhat

On 1 November 2016 at 13:39, Alberto Llamas <albertollam...@gmail.com>
wrote:

> Hello Serhat,
>
> When you are using the webphone (sipml5) by WebRTC the media is secured
> with SRTP. So if the other end-point supports SRTP usually you don't have
> major issues. It is like when you communicate between two sipml5 web phones
> A and B.
>
> But when you are trying to communicate to the IMS softphone, be sure that
> the softphone supports SRTP otherwise you will need to configure a RTP
> Proxy like RTPEngine in kamailio module in order to "translate" between
> plain RTP and SRTP.
>
> This is what I see is your issue based on the pcap files.
>
> PS: You can have a setup in your kamailio config file to offer first SRTP
> and if the other end-point doesn't support it (when you receive a 4XX
> reply) then send a Re-INVITE with plain RTP.
>
> Regards,
>
> On Tue, Nov 1, 2016 at 12:45 PM, Serhat Guler <srtgu...@gmail.com> wrote:
>
>> Hi Daniel, hi Alberto,
>>
>> Thanks for your prompt replies. I have put 2 pcap files in dropbox (
>> https://www.dropbox.com/sh/fzclmbpniebrvx1/AAAOOv4h2ci7bJuuJvSbs3poa?dl=0 )
>> . trace.mercuro.pcap is the one where the session is set up, but there is
>> no audio flow and trace.boghe.pcap is the one with 488 error.
>>
>> Cheers,
>> Serhat
>>
>> On 1 November 2016 at 12:39, Daniel-Constantin Mierla <mico...@gmail.com>
>> wrote:
>>
>>> Hello,
>>>
>>> can you get the SIP INVITE content that was received by the endpoint
>>> returning 488? Maybe we can spot if there is something wrong in the sip
>>> message content or an issue in the endpoint software. Maybe it doesn't like
>>> headers with random string instead of ip addresses (e.g., in via, contact
>>> ...).
>>>
>>> I am not aware of any ims softphone with webrtc capabilities.
>>> Cheers,
>>> Daniel
>>>
>>>
>>> On 01/11/16 12:15, Serhat Guler wrote:
>>>
>>> Hi,
>>>
>>> I have a setup as follows:
>>>
>>> IMS enabled on Kamailio and whereas websockets are enabled for PCSCF for
>>> webrtc calls.
>>>
>>> Calls(both audio and video) between to sipml5 clients using firefox web
>>> browser is possible. The session is setup for the calls from sipml5 to
>>> Mercuro, but then there isn't audio flow as the codecs are not compatible.
>>>
>>> Now I want to test it with Boghe which supports G.722, PCMA, PCMU, and
>>> OPUS codecs as firefox but this time the session isn't being setup. Boghe
>>> replies with "Reason: SIP; cause=488; text="Bad content"
>>> ​" I have seen a similar issue has been mentioned here:
>>> https://github.com/c00lz3r0/boghe/issues/157  but the initial invite
>>> request from sipml5 does have the SDP with media attributes.
>>> ​
>>>
>>> ​Any advice or are there any other IMS softphones that I can use to test
>>> for this scenario. Thanks a lot.
>>>
>>> P.S. The previous email went out directly unintentionally.
>>> Serhat
>>>
>>>
>>>
>>> ___
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>>>
>>>
>>> --
>>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - 
>>> http://www.linkedin.com/in/miconda
>>> Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com
>>>
>>>
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>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>
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>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
>
> --
> Alberto Llamas
> Phone: +1-786-805-6003
> Telecommunications Engineer
> Digium Certified Asterisk Professional (dCap)
>
>
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Re: [SR-Users] Kamailio + IMS + sipml5 call to softphone

[Serhat Guler]

Hi Daniel, hi Alberto,

Thanks for your prompt replies. I have put 2 pcap files in dropbox (
https://www.dropbox.com/sh/fzclmbpniebrvx1/AAAOOv4h2ci7bJuuJvSbs3poa?dl=0 )
. trace.mercuro.pcap is the one where the session is set up, but there is
no audio flow and trace.boghe.pcap is the one with 488 error.

Cheers,
Serhat

On 1 November 2016 at 12:39, Daniel-Constantin Mierla <mico...@gmail.com>
wrote:

> Hello,
>
> can you get the SIP INVITE content that was received by the endpoint
> returning 488? Maybe we can spot if there is something wrong in the sip
> message content or an issue in the endpoint software. Maybe it doesn't like
> headers with random string instead of ip addresses (e.g., in via, contact
> ...).
>
> I am not aware of any ims softphone with webrtc capabilities.
> Cheers,
> Daniel
>
>
> On 01/11/16 12:15, Serhat Guler wrote:
>
> Hi,
>
> I have a setup as follows:
>
> IMS enabled on Kamailio and whereas websockets are enabled for PCSCF for
> webrtc calls.
>
> Calls(both audio and video) between to sipml5 clients using firefox web
> browser is possible. The session is setup for the calls from sipml5 to
> Mercuro, but then there isn't audio flow as the codecs are not compatible.
>
> Now I want to test it with Boghe which supports G.722, PCMA, PCMU, and
> OPUS codecs as firefox but this time the session isn't being setup. Boghe
> replies with "Reason: SIP; cause=488; text="Bad content"
> ​" I have seen a similar issue has been mentioned here:
> https://github.com/c00lz3r0/boghe/issues/157  but the initial invite
> request from sipml5 does have the SDP with media attributes.
> ​
>
> ​Any advice or are there any other IMS softphones that I can use to test
> for this scenario. Thanks a lot.
>
> P.S. The previous email went out directly unintentionally.
> Serhat
>
>
>
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>
>
> --
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> http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com
>
>
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[SR-Users] Fwd: Kamailio + IMS + sipml5 call to softphone

[Serhat Guler]

Hi,

I have a setup as follows:

IMS enabled on Kamailio and whereas websockets are enabled for PCSCF for
webrtc calls.

Calls(both audio and video) between to sipml5 clients using firefox web
browser is possible. The session is setup for the calls from sipml5 to
Mercuro, but then there isn't audio flow as the codecs are not compatible.

Now I want to test it with Boghe which supports G.722, PCMA, PCMU, and OPUS
codecs as firefox but this time the session isn't being setup. Boghe
replies with "Reason: SIP; cause=488; text="Bad content"
​" I have seen a similar issue has been mentioned here:
https://github.com/c00lz3r0/boghe/issues/157  but the initial invite
request from sipml5 does have the SDP with media attributes.
​

​Any advice or are there any other IMS softphones that I can use to test
for this scenario. Thanks a lot.

P.S. The previous email went out directly unintentionally.
Serhat
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[SR-Users] Kamailio + IMS + sipml5 call to softphone

[Serhat Guler]

Hi,

I have a setup as follows:

IMS enabled on Kamailio and whereas websockets are enabled for PCSCF for
webrtc calls.

Calls(both audio and video) between to sipml5 clients using firefox web
browser is possible. The session is setup for the calls from sipml5 to
Mercuro, but then there isn't audio flow as the codecs are not compatible.

Now I want to test it with Boghe which supports G.722, PCMA, PCMU, and OPUS
codecs as firefox but this time the session isn't being setup. Boghe
replies with "Reason: SIP; cause=488; text="Bad content"
​" I have seen a similar issue has been mentioned here:
https://github.com/c00lz3r0/boghe/issues/157  but the initial invite
request from sipml5 does have the SDP with media attributes.
​
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Re: [SR-Users] PCSCF cannot create via header for sipml5 ACK package

[Serhat Guler]

Hi Alberto,

Removing the outbound proxy solved the problem. Thanks for your help.

Cheers,
Serhat

On 30 October 2016 at 10:52, Alberto Llamas <albertollam...@gmail.com>
wrote:

> Hi Serhat,
>
> I am not sure how is the setup of your network, but you should remove the
> outbound proxy setting from sipml5 (SIP outbound Proxy URL: udp://
> 192.168.0.11:4060).
>
> Test it and let us know.
>
> Regards,
>
> On Sat, Oct 29, 2016 at 9:38 PM, Serhat Guler <srtgu...@gmail.com> wrote:
>
>> Hi all,
>>
>> I am still stuck with the ACK message not being forwarded by the
>> originating PCSCF. Any advice would be great.
>>
>> Thanks,
>> Serhat
>>
>> On 24 October 2016 at 21:00, Serhat Guler <srtgu...@gmail.com> wrote:
>>
>>> Hi Daniel,
>>>
>>> I am using only record_route() without any parameters. I do not have a
>>> proper computer atm to draw the network diagram, but I can tell you shortly
>>> about the network setup.
>>>
>>> I have only enabled websockets for the pcscf to allow ws and wss
>>> connections. In that case there is a ws connection that uses UDP protocol.
>>> This is the ACK to complete the session setup.
>>>
>>> the sipml5 client is configured as follows:
>>> WebSocket Server URL: ws://192.168.0.11:880
>>> SIP outbound Proxy URL: udp://192.168.0.11:4060
>>>
>>> Mercuro IMS client: uses UDP port as well: 4060
>>>
>>> The call is made from sipml5 client. The Mercuro phone rings, and when I
>>> reply the call, 200 OK is sent to sipml5 webrtc client, but the ACK from
>>> sipml5 doesn't pass the PCSCF as I explained in the previous message.
>>>
>>> A part of PCSCF cfg file:
>>>
>>> # Check for Subsequent requests:
>>> if (has_totag()) {
>>> # sequential request withing a dialog should
>>> # take the path determined by record-routing
>>> if (loose_route()) {
>>> if ($route_uri =~ "sip:mo@.*") {
>>> setflag(FLT_MO);
>>> }
>>> if(!isdsturiset()) {
>>> handle_ruri_alias();
>>> }
>>> # RTP-Relay, if necessary
>>> route(RTPPROXY);
>>> t_relay();
>>> } else {
>>> if ( is_method("ACK") ) {
>>> if ( t_check_trans() ) {
>>> # no loose-route, but stateful ACK;
>>> # must be an ACK after a 487
>>> # or e.g. 404 from upstream server
>>> t_relay();
>>> exit;
>>> } else {
>>> xlog("L_INFO", "ACK without matching transaction ...
>>> ignore and discard!\n");
>>> # ACK without matching transaction ... ignore and
>>> discard
>>> exit;
>>> }
>>> }
>>> sl_send_reply("404","Not here");
>>> }
>>> exit;
>>> }
>>>
>>> Cheers,
>>> Serhat
>>>
>>>
>>>
>>> On 24 October 2016 at 20:18, Daniel-Constantin Mierla <mico...@gmail.com
>>> > wrote:
>>>
>>>> Hello,
>>>>
>>>> I haven't noticed the log files, it's ok.
>>>>
>>>> From the Route header, I see that there is a proxy that uses WS:
>>>>
>>>> Route: <sip:mo@192.168.0.11:880;transport=ws;r2=on;lr=on;ftag=GxzKy
>>>> 1nCMEI1mR0RztrB;did=e82.0c3>
>>>> That is the address of the next hop and typically a proxy doesn't use
>>>> websocket connection to another proxy. Can you show a diagram with the sip
>>>> server nodes in your network and what protocols are used between them?
>>>>
>>>> Are you simply use record_route() function, or some other function or
>>>> different parameters to it?
>>>>
>>>> Cheers,
>>>> Daniel
>>>>
>>>>
>>>> On 24/10/16 12:18, Serhat Guler wrote:
>>>>
>>>> Hi Daniel,
>>>>
>>>> Thanks for your reply. I actually attached a log file with debug level
>>>> 3, consisting ACK related messages. If you would like to see more logs,
>>>> I'll send a new log file in the evening.
>>>>

Re: [SR-Users] PCSCF cannot create via header for sipml5 ACK package

[Serhat Guler]

Hi all,

I am still stuck with the ACK message not being forwarded by the
originating PCSCF. Any advice would be great.

Thanks,
Serhat

On 24 October 2016 at 21:00, Serhat Guler <srtgu...@gmail.com> wrote:

> Hi Daniel,
>
> I am using only record_route() without any parameters. I do not have a
> proper computer atm to draw the network diagram, but I can tell you shortly
> about the network setup.
>
> I have only enabled websockets for the pcscf to allow ws and wss
> connections. In that case there is a ws connection that uses UDP protocol.
> This is the ACK to complete the session setup.
>
> the sipml5 client is configured as follows:
> WebSocket Server URL: ws://192.168.0.11:880
> SIP outbound Proxy URL: udp://192.168.0.11:4060
>
> Mercuro IMS client: uses UDP port as well: 4060
>
> The call is made from sipml5 client. The Mercuro phone rings, and when I
> reply the call, 200 OK is sent to sipml5 webrtc client, but the ACK from
> sipml5 doesn't pass the PCSCF as I explained in the previous message.
>
> A part of PCSCF cfg file:
>
> # Check for Subsequent requests:
> if (has_totag()) {
> # sequential request withing a dialog should
> # take the path determined by record-routing
> if (loose_route()) {
> if ($route_uri =~ "sip:mo@.*") {
> setflag(FLT_MO);
> }
> if(!isdsturiset()) {
> handle_ruri_alias();
> }
> # RTP-Relay, if necessary
> route(RTPPROXY);
> t_relay();
> } else {
> if ( is_method("ACK") ) {
> if ( t_check_trans() ) {
> # no loose-route, but stateful ACK;
> # must be an ACK after a 487
> # or e.g. 404 from upstream server
> t_relay();
> exit;
> } else {
> xlog("L_INFO", "ACK without matching transaction ...
> ignore and discard!\n");
> # ACK without matching transaction ... ignore and
> discard
> exit;
> }
> }
> sl_send_reply("404","Not here");
> }
> exit;
> }
>
> Cheers,
> Serhat
>
>
>
> On 24 October 2016 at 20:18, Daniel-Constantin Mierla <mico...@gmail.com>
> wrote:
>
>> Hello,
>>
>> I haven't noticed the log files, it's ok.
>>
>> From the Route header, I see that there is a proxy that uses WS:
>>
>> Route: <sip:mo@192.168.0.11:880;transport=ws;r2=on;lr=on;ftag=GxzKy
>> 1nCMEI1mR0RztrB;did=e82.0c3>
>> That is the address of the next hop and typically a proxy doesn't use
>> websocket connection to another proxy. Can you show a diagram with the sip
>> server nodes in your network and what protocols are used between them?
>>
>> Are you simply use record_route() function, or some other function or
>> different parameters to it?
>>
>> Cheers,
>> Daniel
>>
>>
>> On 24/10/16 12:18, Serhat Guler wrote:
>>
>> Hi Daniel,
>>
>> Thanks for your reply. I actually attached a log file with debug level 3,
>> consisting ACK related messages. If you would like to see more logs, I'll
>> send a new log file in the evening.
>>
>> Cheers,
>> Serhat
>>
>> On 24 October 2016 at 12:13, Daniel-Constantin Mierla <mico...@gmail.com>
>> wrote:
>>
>>> Hello,
>>>
>>> can you get all the log messages for ACK but with debug=3 in the
>>> kamailio.cfg?
>>>
>>> Cheers,
>>> Daniel
>>>
>>> On 23/10/16 22:04, Serhat Guler wrote:
>>>
>>> ​Hello,
>>>
>>> I finally managed to place a call from sipml5 webrtc client​ to Mercuro
>>> IMS client. The phone rings, and when I answer it sends 200 OK to the
>>> sipml5 where as sipml5 send back an ACK message which never passes the
>>> originating PCSCF. The PCSCF says:
>>>
>>>  8(3640) WARNING:  [msg_translator.c:2729]: via_builder(): TCP/TLS
>>> connection (id: 0) for WebSocket could not be found
>>>  8(3640) ERROR:  [msg_translator.c:1947]:
>>> build_req_buf_from_sip_req(): could not create Via header
>>>  8(3640) ERROR:  [forward.c:548]: forward_request(): building
>>> failed
>>>
>>> I doubt that the WebSocket connection is closed, cause when I terminate
>>> the call from Mercuro client a bye request is being sent to the sipml5.
>>>
>>> The 

Re: [SR-Users] PCSCF cannot create via header for sipml5 ACK package

[Serhat Guler]

Hi Daniel,

Thanks for your reply. I actually attached a log file with debug level 3,
consisting ACK related messages. If you would like to see more logs, I'll
send a new log file in the evening.

Cheers,
Serhat

On 24 October 2016 at 12:13, Daniel-Constantin Mierla <mico...@gmail.com>
wrote:

> Hello,
>
> can you get all the log messages for ACK but with debug=3 in the
> kamailio.cfg?
>
> Cheers,
> Daniel
>
> On 23/10/16 22:04, Serhat Guler wrote:
>
> ​Hello,
>
> I finally managed to place a call from sipml5 webrtc client​ to Mercuro
> IMS client. The phone rings, and when I answer it sends 200 OK to the
> sipml5 where as sipml5 send back an ACK message which never passes the
> originating PCSCF. The PCSCF says:
>
>  8(3640) WARNING:  [msg_translator.c:2729]: via_builder(): TCP/TLS
> connection (id: 0) for WebSocket could not be found
>  8(3640) ERROR:  [msg_translator.c:1947]:
> build_req_buf_from_sip_req(): could not create Via header
>  8(3640) ERROR:  [forward.c:548]: forward_request(): building failed
>
> I doubt that the WebSocket connection is closed, cause when I terminate
> the call from Mercuro client a bye request is being sent to the sipml5.
>
> The ACK package:
>
> ACK sip:alice@192.168.0.10:49794;transport=udp SIP/2.
> Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKvuly7bmxnN4aqM4zZTIS;
> rport
> From: "Bob"<sip:b...@net1.test>;tag=GxzKy1nCMEI1mR0RztrB
> To: <sip:al...@net1.test>;tag=18823
> Contact: "Bob"<sip:bob@df7jal23ls0d.invalid;rtcweb-breaker=no;
> click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
> Call-ID: 5a500969-d0fa-14d1-7d0e-8605f4356ca6
> CSeq: 3887 ACK
> Content-Length:
> Route: <sip:192.168.0.11:4060;lr;sipml5-outbound;transport=udp>
> Max-Forwards: 69
> Route: <sip:mo@192.168.0.11:880;transport=ws;r2=on;lr=on;ftag=
> GxzKy1nCMEI1mR0RztrB;did=e82.0c3>
> Route: <sip:mo@192.168.0.11:4060;r2=on;lr=on;ftag=
> GxzKy1nCMEI1mR0RztrB;did=e82.0c3>
> Route: <sip:mo@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;
> did=e82.f062>
> Route: <sip:mt@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;
> did=e82.f062>
> Route: <sip:mt@192.168.0.11:4060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;
> did=e82.1c3>
> User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
> Organization: Doubango Telecom
>
> Have been thinking for quite a while, but couldn't really find a reason
> why it wouldn't add the v,a header. A debug 3 level log file is also
> attached.
>
> Thanks in advance,
> Serhat
>
>
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>
> --
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> http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com
>
>
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[SR-Users] PCSCF cannot create via header for sipml5 ACK package

[Serhat Guler]

​Hello,

I finally managed to place a call from sipml5 webrtc client​ to Mercuro IMS
client. The phone rings, and when I answer it sends 200 OK to the sipml5
where as sipml5 send back an ACK message which never passes the originating
PCSCF. The PCSCF says:

 8(3640) WARNING:  [msg_translator.c:2729]: via_builder(): TCP/TLS
connection (id: 0) for WebSocket could not be found
 8(3640) ERROR:  [msg_translator.c:1947]:
build_req_buf_from_sip_req(): could not create Via header
 8(3640) ERROR:  [forward.c:548]: forward_request(): building failed

I doubt that the WebSocket connection is closed, cause when I terminate the
call from Mercuro client a bye request is being sent to the sipml5.

The ACK package:

ACK sip:alice@192.168.0.10:49794;transport=udp SIP/2.
Via: SIP/2.0/WS
df7jal23ls0d.invalid;branch=z9hG4bKvuly7bmxnN4aqM4zZTIS;rport
From: "Bob";tag=GxzKy1nCMEI1mR0RztrB
To: ;tag=18823
Contact: "Bob";+g.oma.sip-im;language="en,fr"
Call-ID: 5a500969-d0fa-14d1-7d0e-8605f4356ca6
CSeq: 3887 ACK
Content-Length:
Route: 
Max-Forwards: 69
Route: 
Route: 
Route: 
Route: 
Route: 
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom

Have been thinking for quite a while, but couldn't really find a reason why
it wouldn't add the v,a header. A debug 3 level log file is also attached.

Thanks in advance,
Serhat


debug3.log
Description: Binary data
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Re: [SR-Users] sipml5 fail invite kamailio ims web sockets

[Serhat Guler]

Hi Daniel,

Thank you for your reply. I'll defo check it out in the evening. Do you
think the scenario I explained here (http://lists.sip-router.org/
pipermail/sr-users/2016-October/094831.html) might also be the case ?

Cheers,
Serhat

On 20 October 2016 at 11:22, Daniel-Constantin Mierla <mico...@gmail.com>
wrote:

> Hello,
>
> it seems that the response to the invite doesn't have a Contact header.
> Can you check that? You can print the message in logs with $mb.
> Cheers,
> Daniel
>
>
> On 17/10/16 22:46, Serhat Guler wrote:
>
> Dear all,
>
> I can register the sipml5 client to my kamailio IMS setup successfully,
> but when I try to make a call the invite fails with "478 Unresolvable
> destination". From the wireshark trace file I think the PCSCF on the
> terminating side cannot resolve the destination. After the invite request
> fails the caller re-registers to the networkj automatically. The wireshark
> file is attached.
>
> The registration process happens successfully but all of the nodes give
> some errors:
>
> 10(3773) INFO: 

[SR-Users] Terminating PCSCF unresolvable request URI (Kamailio IMS Websockets)

[Serhat Guler]

Dear all,

I am trying to make a call between 2 webrtc clients(sipml5) using my
kamailio IMS setup. The invite request fails at the terminating PCSCF with
error code "Status: 478 Unresolvable destination (478/TM)". I have traced
the packages and saw that the package that arrives at the terminating PCSCF
has an unresolvable request URI due to the websocket connection.

The package being sent from terminating ICSCF to SCSCF has the
"Request-URI: sip:b...@net1.test"
The package from terminating SCSCF to terminating PCSCF has the "Request-URI:
sip:bob@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws"

I guess the terminating SCSCF replaces the request URI with the one it
stored during the registration process, but the terminating PCSCF cannot
resolve the host name number; therefore, it reports back "Unresolvable
destination". Is that really the case happening in here ? How can I get
over this problem ? Any ideas are welcome.

Cheers,
Serhat
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[SR-Users] sipml5 fail invite kamailio ims web sockets

[Serhat Guler]

Dear all,

I can register the sipml5 client to my kamailio IMS setup successfully, but
when I try to make a call the invite fails with "478 Unresolvable
destination". From the wireshark trace file I think the PCSCF on the
terminating side cannot resolve the destination. After the invite request
fails the caller re-registers to the networkj automatically. The wireshark
file is attached.

The registration process happens successfully but all of the nodes give
some errors:

10(3773) INFO: 

Re: [SR-Users] sipml5 register to kamailio IMS (Unauthorized - Challenging the UE)

[Serhat Guler]

Hi Daniel,

Thanks for your reply. I dug in a bit more and found out that the sipml5
doesn't calculate the digest correctly when AKAv1MD5 or v2 is selected as
default (have no idea why). So, I set the default authentication algorithm
as Digest MD5 and now I can register the UE.

Cheers,
Serhat

On 17 October 2016 at 11:47, Daniel-Constantin Mierla <mico...@gmail.com>
wrote:

> Hello,
>
> maybe you can get more hints about what happens there by running with
> debug=3 inside the kamailio cfg.
>
> Cheers,
> Daniel
>
> On 14/10/16 17:40, Serhat Guler wrote:
>
> Hi,
>
> I am trying to register sipml5 webrtc client to my kamailio IMS setup. I
> have tried to register the client both with ws and wss, but it seems to be
> that the sipml5 doesn't calculate the authentication digest right. The
> authentication mechanism i set to AKAv1-MD5 as default in the hss. A simple
> wireshark file is attached. .10 being the host, .11 being the kamailio
> server.
>
> From the output of scscf we can see that the digests do not match.
>
> [REGISTER] from [sip:b...@net1.test] to [sip:b...@net1.test]
> ims_auth [authorize.c:824]: authenticate(): uri=sip:net1.test
> nonce=rB2iDuerHwoy+LUStSOsYojAESfWmAAApFZ3XNB8FdA= response=
> 61cbdbeb47c9880ededfca51c3801800 qop=auth-int nc=0001 cnonce=
> 2d4545cf4c935c8094c8b1da3d4a2976 hbody=d41d8cd98f00b204e9800998ecf8427e
> ims_auth [authorize.c:872]: authenticate(): UE said:
> 61cbdbeb47c9880ededfca51c3801800 and we expect
> 7e27ef414cf37d96cd1c849bb7e59415 ha1 a53988ba0b257941bc747b1026225c77
> (REGISTER)
> tm [tm.c:1265]: w_t_reply(): ERROR: t_reply: cannot send a t_reply to a
> message for which no T-state has been established
>
> Thanks a lot,
> Serhat
>
>
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>
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> http://www.linkedin.com/in/miconda
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>
>
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[SR-Users] sipml5 register to kamailio IMS (Unauthorized - Challenging the UE)

[Serhat Guler]

Hi,

I am trying to register sipml5 webrtc client to my kamailio IMS setup. I
have tried to register the client both with ws and wss, but it seems to be
that the sipml5 doesn't calculate the authentication digest right. The
authentication mechanism i set to AKAv1-MD5 as default in the hss. A simple
wireshark file is attached. .10 being the host, .11 being the kamailio
server.

>From the output of scscf we can see that the digests do not match.

[REGISTER] from [sip:b...@net1.test] to [sip:b...@net1.test]
ims_auth [authorize.c:824]: authenticate(): uri=sip:net1.test
nonce=rB2iDuerHwoy+LUStSOsYojAESfWmAAApFZ3XNB8FdA=
response=61cbdbeb47c9880ededfca51c3801800 qop=auth-int nc=0001
cnonce=2d4545cf4c935c8094c8b1da3d4a2976
hbody=d41d8cd98f00b204e9800998ecf8427e
ims_auth [authorize.c:872]: authenticate(): UE said:
61cbdbeb47c9880ededfca51c3801800 and we expect
7e27ef414cf37d96cd1c849bb7e59415 ha1 a53988ba0b257941bc747b1026225c77
(REGISTER)
tm [tm.c:1265]: w_t_reply(): ERROR: t_reply: cannot send a t_reply to a
message for which no T-state has been established

Thanks a lot,
Serhat


jssip.ws.pcap
Description: Binary data
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[SR-Users] Websocket with kamailio IMS

[Serhat Guler]

Dear all,

I am trying to configure websockets for kamailio IMS. My basic
kamailio(without IMS) setup works properly with websockets and without
websockets. The IMS functionality also works, but only with IMS clients and
now I want to use webrtc clients and make calls. At the moment I am trying
to add the websockets support to the pcscf. I have did some changes to the
pcscf cfg file.

My problem is not really related to the cfg of pcscf as the component is
listening for tcp/tls connections, but I cannot connect through the
assigned ports. The output of the pcscf component and the the wireshark
pcap file for the Websocket are attached. (I basically tried to browse
http://server.ip.address:tcp.port or https://server.ip:tls.port without
using a webrtc client yet).

192.168.0.10 -> host
192.168.0.11 -> virtual machine running kamailio

Thanks in advance,
Serhat
Listening on
 udp: server.net1.test [192.168.0.11]:4060
 tcp: 192.168.0.11 [192.168.0.11]:880
 tcp: server.net1.test [192.168.0.11]:4060
 tls: 192.168.0.11 [192.168.0.11]:4443
 tls: server.net1.test [192.168.0.11]:5061
Aliases:
 *: pcscf.net1.test:*

WARNING: no fork mode
 0(1537) INFO:  [tcp_main.c:4665]: init_tcp(): using epoll_lt as the io 
watch method (auto detected)
 0(1537) INFO: rr [../outbound/api.h:54]: ob_load_api(): unable to import 
bind_ob - maybe module is not loaded
 0(1537) INFO: rr [rr_mod.c:174]: mod_init(): outbound module not available
 0(1537) INFO: ims_usrloc_pcscf [hslot.c:62]: ul_init_locks(): locks array size 
512
 0(1537) INFO: tls [tls_mod.c:362]: mod_init(): With ECDH-Support!
 0(1537) INFO: tls [tls_mod.c:365]: mod_init(): With Diffie Hellman
 0(1537) INFO: tls [tls_init.c:593]: init_tls_h(): tls: _init_tls_h:  compiled  
with  openssl  version "OpenSSL 1.0.1t  3 May 2016" (0x1000114f), kerberos 
support: off, compression: on
 0(1537) INFO: tls [tls_init.c:601]: init_tls_h(): tls: init_tls_h: installed 
openssl library version "OpenSSL 1.0.1t  3 May 2016" (0x1000114f), kerberos 
support: off,  zlib compression: off
 compiler: gcc -I. -I.. -I../include  -fPIC -DOPENSSL_PIC -DOPENSSL_THREADS 
-D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -DL_ENDIAN -DTERMIO -g -O2 
-fstack-protector-strong -Wformat -Werror=format-security -D_FORTIFY_SOURCE=2 
-Wl,-z,relro -Wa,--noexecstack -Wall -march=i686 -DOPENSSL_BN_ASM_PART_WORDS 
-DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_GF2m -DSHA1_ASM 
-DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DRMD160_ASM -DAES_ASM -DVPAES_ASM 
-DWHIRLPOOL_ASM -DGHASH_ASM
 0(1537) WARNING: tls [tls_init.c:655]: init_tls_h(): tls: openssl bug #1491 
(crash/mem leaks on low memory) workaround enabled (on low memory tls 
operations will fail preemptively) with free memory thresholds 8388608 and 
4194304 bytes
 0(1537) INFO:  [cfg/cfg_ctx.c:608]: cfg_set_now(): INFO: cfg_set_now(): 
tls.low_mem_threshold1 has been changed to 8388608
 0(1537) INFO:  [cfg/cfg_ctx.c:608]: cfg_set_now(): INFO: cfg_set_now(): 
tls.low_mem_threshold2 has been changed to 4194304
 0(1537) INFO:  [udp_server.c:150]: probe_max_receive_buffer(): SO_RCVBUF 
is initially 163840
 0(1537) INFO:  [udp_server.c:200]: probe_max_receive_buffer(): SO_RCVBUF 
is finally 327680
 0(1537) INFO: tls [tls_domain.c:276]: fill_missing(): TLSs: 
tls_method=3
 0(1537) INFO: tls [tls_domain.c:288]: fill_missing(): TLSs: 
certificate='/usr/local/etc/kamailio/kamailio-selfsigned.pem'
 0(1537) INFO: tls [tls_domain.c:295]: fill_missing(): TLSs: 
ca_list='/usr/local/src/kamailio-4.3/kamailio/etc/tls/rootCA/cacert.pem'
 0(1537) INFO: tls [tls_domain.c:302]: fill_missing(): TLSs: 
crl='(null)'
 0(1537) INFO: tls [tls_domain.c:306]: fill_missing(): TLSs: 
require_certificate=0
 0(1537) INFO: tls [tls_domain.c:313]: fill_missing(): TLSs: 
cipher_list='(null)'
 0(1537) INFO: tls [tls_domain.c:320]: fill_missing(): TLSs: 
private_key='/usr/local/etc/kamailio/kamailio-selfsigned.key'
 0(1537) INFO: tls [tls_domain.c:324]: fill_missing(): TLSs: 
verify_certificate=0
 0(1537) INFO: tls [tls_domain.c:327]: fill_missing(): TLSs: 
verify_depth=9
 0(1537) INFO: tls [tls_domain.c:671]: set_verification(): TLSs: No 
client certificate required and no checks performed
 0(1537) INFO: tls [tls_domain.c:276]: fill_missing(): TLSc: 
tls_method=12
 0(1537) INFO: tls [tls_domain.c:288]: fill_missing(): TLSc: 
certificate='(null)'
 0(1537) INFO: tls [tls_domain.c:295]: fill_missing(): TLSc: 
ca_list='(null)'
 0(1537) INFO: tls [tls_domain.c:302]: fill_missing(): TLSc: 
crl='(null)'
 0(1537) INFO: tls [tls_domain.c:306]: fill_missing(): TLSc: 
require_certificate=1
 0(1537) INFO: tls [tls_domain.c:313]: fill_missing(): TLSc: 
cipher_list='(null)'
 0(1537) INFO: tls [tls_domain.c:320]: fill_missing(): TLSc: 
private_key='(null)'
 0(1537) INFO: tls [tls_domain.c:324]: fill_missing(): TLSc: 
verify_certificate=1
 0(1537) INFO: tls [tls_domain.c:327]: fill_missing(): TLSc: 
verify_depth=9
 0(1537) INFO: tls