Re: [SR-Users] Status of db_cassandra

[Andreas Granig]

Hi,

I also believe the module needs to be marked somehow, because it's
implemented against a very old cassandra version and does not work with
recent versions at all.

I'm afraid a complete re-implementation from scratch is needed there,
because the protocol changed significantly. Also schema definitions for
cassandra are not aligned anymore with the current usrloc schema, so all
in all it's completely broken last time I checked (~6 months ago).

I was briefly in touch with the original developers back then, and the
feedback was that it was a proof-of-concept and never got put somewhere
in production, so we can safely assume no-one is using it anyways (and
it won't work either).

Andreas

On 03/21/2017 04:24 PM, Markus Bönke wrote:
> Hello Daniel,
> 
> to change the status to „unmaintained“ sounds OK for me.
> 
> Regards,
> 
> Markus
> 
>> Am 21.03.2017 um 13:53 schrieb Daniel-Constantin Mierla
>> >:
>>
>> Hello,
>>
>> having some interest on the module and cassandra being still actual, I
>> tried to avoid moving it as deprecated/obsoleted module as maybe
>> someone will just pick it and do the required updates. It may still
>> work for older versions of kamailio and cassandra, so maybe a better
>> tag for it will be 'unmaintained' or 'not-up-to-date', to express more
>> accurate the status. I consider using deprecated/obsoleted when there
>> is reason to keep the module at all.
>>
>> Cheers,
>> Daniel
>>
>> On Tue, Mar 21, 2017 at 1:37 PM, Markus Bönke > > wrote:
>>
>> Hello Daniel,
>>
>> thanks for the info. Maybe it’s better to put the module into
>> status “Deprecated“? 
>>
>> Regards
>>
>> Markus
>>> Am 21.03.2017 um 11:53 schrieb Daniel-Constantin Mierla
>>> >:
>>>
>>> Hello,
>>>
>>> unfortunately the module db_cassandra was not really maintained
>>> and it has been reported to have issues even at start up. No one
>>> has picked it up yet to get it up to date, hopefully someone will
>>> do it at some point. I don't use and I don't have any access to a
>>> testbed with cassandra, so I was not able to assist with it.
>>>
>>> db_mongodb should work from the no-SQL db connectors we have.
>>>
>>> Cheers,
>>>
>>> Daniel
>>>
>>> March 21, 2017 9:56 AM, "Markus Bönke" >> > wrote:
>>>
>>> Hello,
>>> We are thinking about to use kamailio as a sip registrar with
>>> cassandra as db backend. As I can see in the documentation
>>> for kamailio 5.0 it is only tested with Casandra 1.1.6 and
>>> 1.0.1, current version is 3.10 in the meantime. Is anyone
>>> using this module with newer versions of Cassandra? Is it
>>> stable?
>>> Thanks and regards
>>> Markus
>>>
>>> __
>>>
>>> --
>>> Daniel-Constantin Mierla
>>> www.kamailio.org  -- www.asipto.com
>>> __
>>> ___
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>>> 
>>
>>
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>>
>>
>> -- 
>> Daniel-Constantin Mierla - http://www.asipto.com 
>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/micond
>> 
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Re: [SR-Users] Rate limiting

[Andreas Granig]

Hi Alex,

http://www.kamailio.org/docs/modules/4.4.x/modules/ratelimit.html is
another approach here. Tried it as a proof of concept recently and it
seems to do its job.

Seems like pipelimit is derived from ratelimit. What's the main
difference from ratelimit, other than named pipes and DB support? What's
the purpose of the DB support of pipelimit? Does it cache its values and
can be reloaded from DB on demand (I don't see an rpc command for that)?
That would be really valuable.

Andreas

On 08/29/2016 05:39 PM, Alex Balashov wrote:
> On 08/29/2016 11:37 AM, NITESH BANSAL wrote:
> 
>> Finally I got it working. The issue was that I was trying to use
>> pikelimit with Kamailio version 4.1, 4.1 version doesn't allow for
>> dynamic pipe creation.
>>
>> In the end, I backported pipelimit code from Kamailio version 4.2 and
>> used pl_check function to create dynamic pipes.
> 
> Excellent. That was indeed an important shift. :-)
> 

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Re: [SR-Users] db_cassandra status and plans

[Andreas Granig]

Hi Lucian,

On 03/31/2016 01:36 PM, Lucian Balaceanu wrote:
> At 1&1 we actually haven't used db_cassandra, but were aware of some of
> its shortcomings (e.g. thrift interface).
> Do you consider Cassandra is stable and easy enough to maintain so as to
> use it as usrloc back-end? Are you
> also considering other alternatives?

We're planning to introduce cassandra for sharding of certain huge data
sets (voice/video-mails etc.), and related to kamailio maybe using it to
distribute heavy writes in a cluster (to sync location or certain
presence information) is at least under serious consideration, but
really depends on benchmarking results. So for now cassandra is still in
evaluation.

For usrloc itself, we used to use a quite old mysql-cluster (ndb)
version with good success for years, but stability during maintenance
has always been a concern, as it was quite fragile if you touched it
(extending nodes etc). This, and performance, seems to have been
improved a lot over the last years, so we're reconsidering that as
another option at least for kamailio-related parts. Advantage here is
that it just works transparently from a mysql driver point of view.

Andreas

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Re: [SR-Users] db_cassandra status and plans

[Andreas Granig]

Hi Daniel,

On 03/31/2016 12:45 PM, Daniel-Constantin Mierla wrote:
> I haven't used Cassandra at all, maybe some of the devs at 1&1 can share
> if they have any plans or usage stats for it, being the initial
> developers of the module.

Let's see if someone chimes in. Not sure if Anca is still involved, or
if/how it's actually used at 1&1 or anywhere else.

> For me it is ok to introduce a new module as well, the old one can be
> removed afterwards if the new one overtakes it in features.

Makes sense, although I doubt someone is using it with recent Kamailio
versions due to the issues at least I encountered with it.
The new module in itself will be a drop-in replacement of the current
one (so we could re-use the same name but rewriting it from scratch),
the only limitation I can see for now is that it won't support cassandra
version older than 1.2 due to lack of cql support. So having a grace
period (or a parallel module with a different name) is fine with me.

> What I actually wanted to discuss is regarding the query for NAT pinging
> -- with recent versions, there should be a dedicated column for querying
> natted records. It is no longer relying of flags (and bitwise
> operations) for this case. Or did you have in mind something else with:
> 
> """
> The module can not be used together with nathelper for nat pings, since
> the queries don't provide a key, which is required for cassandra.
> """

Actually one of the issues with the cassandra schema is that the new
columns introduced in location e.g. for natping are not reflected in the
module's table schema. I haven't looked into the details of the db query
the module tries to do in case of natping, I was just observing the
errors the module was throwing for each natping attempt.

The plan is to use the new way of querying records for natping anyways.

Andreas

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[SR-Users] db_cassandra status and plans

[Andreas Granig]

Hi all,

I was checking db_cassandra with Kamailio 4.3 recently to try making it
work as a usrloc back-end, and my observation is that the module has
quite some issues.

In general, the module states that it's only tested with cassandra
1.0.6, which is really out-dated. I tried with the latest version of the
1.0 series (1.0.12) and even with lots of fiddling I didn't manage to
make it work, could be my fault though. There has not been any work for
4 years other than cosmetical documentation fixes two years ago.

Documentation is missing important information how to set up the version
table, that could be fixed though. The schema definition in the
documentation is missing new columns which have been introduced, that
could be fixed as well.

The module uses the legacy thrift interface, whereas nowadays (since
cassandra 1.2+) a native cql driver is available as open-source.

The module can not be used together with nathelper for nat pings, since
the queries don't provide a key, which is required for cassandra.

Our plan is to revamp the whole module and make it work with recent
cassandra versions and remove the nat ping limitations. Main question is
whether to go for a new module or replacing the old one, since it
doesn't seem to work anymore anyways.

Feedback whether you got it working with Kamailio 4.3+ and your views on
future plans is much appreciated!

Andreas

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Re: [SR-Users] rtpengine and serial forking with multiple network interfaces

[Andreas Granig]

Hi,

On 12/10/2015 01:33 PM, Alex Balashov wrote:
> I tested an approach in which all initial rtpengine_manage() calls (in
> support of a new SDP offer) were made in a branch_route[], and this
> appeared to work.
> 
> Is this the 'orthodox' way to do it?

That's the way we're using it also.

Andreas

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Re: [SR-Users] SCA fails to update appearance

[Andreas Granig]

Hello Jorj,

On 03/03/2015 06:25 PM, Jorj Bauer wrote:
 One thing I can see is that neither of the devices send a SUBSCRIBE with
 line-seize Event. They rather just subscribe to the call-info event, and
 in the oubound INVITE, they don't send a Call-Info header either. Is
 this mandatory for the SCA module to work?
 
 It sounds like they're not configured for SCA properly.

You're indeed right. Yealink needs the Broadsoft-specific firmware in
order to work with the sca module, otherwise it won't subscribe to
line-seize events.

I think I somewhat figured it out for now and it's basically working.

One thing which is nice but was somewhat unexpected (because it wasn't
documented in the module docs) is that sca_call_info_update() also
handles barge-in when a call is publicly held, so it transforms an
INVITE with the Call-Info header into a message included Replaces and all.

Thanks again,
Andreas

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Re: [SR-Users] SCA fails to update appearance

[Andreas Granig]

Hi Jori,

On 03/02/2015 02:47 PM, Jorj Bauer wrote:
 User pho...@domain.org is registered on two devices A and B
 
 kamcmd sca.all_subscriptions
 phone1 10.15.20.174:5100 call-info 2697 active

I had a mixed setup in my first test with one device registering as
private line and another one as shared line, so I redid the setup to
have two shared lines, now I have two entries.

 Mar  2 12:02:33 sp2 proxy[34517]: WARNING: sca [sca_appearance.c:976]:
 sca_appearance_update_index(): Cannot update sip:pho...@domain.org index
 0 to unknown: index 0 not in use
 Mar  2 12:02:33 sp2 proxy[34517]: ERROR: sca [sca_call_info.c:1046]:
 sca_call_info_invite_request_handler(): Failed to update
 sip:pho...@domain.org appearance-index 0 to unknown
 
 Something is also very wrong here, because unknown is either the 
 uninitialized or an otherwise invalid state.
 
 What kind of devices are these? Can you grab a packet capture, from the first 
 REGISTER to all the way through that call?

With two shared lines (one Yealink T-28 and one Yealink T-22) I'm still
getting the same error.

One thing I can see is that neither of the devices send a SUBSCRIBE with
line-seize Event. They rather just subscribe to the call-info event, and
in the oubound INVITE, they don't send a Call-Info header either. Is
this mandatory for the SCA module to work?

Can I send you a pcap in private?

Thanks,
Andreas

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[SR-Users] SCA fails to update appearance

[Andreas Granig]

Hi guys,

Playing with SCA for the first time, I'm hittin the following issue:

User pho...@domain.org is registered on two devices A and B, and device
B sends SUBSCRIBE to pho...@domain.org with call-info Event.
Subscription is handled properly, and my DB content is this:

id: 4
subscriber: sip:phone1@10.15.20.174:5100;alias=1.2.3.4~5100~1
   aor: sip:pho...@domain.org
 event: 1
   expires: 1425296906
 state: 0
   app_idx: 0
   call_id: dacf5194-3d6a8e16d03338102a940080f0e9071c@10.15.20.174
  from_tag: 4247246264
to_tag: f3067022b00c564156251ba2f28f331f-7c92
  record_route:
sip:127.0.0.1;r2=on;lr=on;ftag=4247246264;nat=yes;ngcplb=yes;socket=udp:1.2.3.4:5060,sip:1.2.3.4;r2=on;lr=on;ftag=4247246264;nat=yes;ngcplb=yes;socket=udp:1.2.3.4:5060
   notify_cseq: 3
subscribe_cseq: 2

Kamcmd reports this:

kamcmd sca.all_subscriptions
phone1 10.15.20.174:5100 call-info 2697 active

If pho...@domain.org on device A does a call, I get this in my logs when
calling sca_call_info_update() right as first thing in my routing config
handling the INVITE:

Mar  2 12:02:33 sp2 proxy[34517]: WARNING: sca [sca_appearance.c:976]:
sca_appearance_update_index(): Cannot update sip:pho...@domain.org index
0 to unknown: index 0 not in use
Mar  2 12:02:33 sp2 proxy[34517]: ERROR: sca [sca_call_info.c:1046]:
sca_call_info_invite_request_handler(): Failed to update
sip:pho...@domain.org appearance-index 0 to unknown

Is there something I'm obviously doing wrong here?

Andreas

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Re: [SR-Users] rtpengine symmetric RTP behavior

[Andreas Granig]

Hi,

On 08/11/2014 08:34 PM, Narsay, Deep wrote:
 Is there any way rtpengine can be configured to use same UDP port to
 receive and transmit RTP  packets?
 
 The set up I'm trying is 
 
 SIP_Client --  Kamailio/rtpengine  --  Freeswitch 
 InternetInternet LAN  LAN 192.168.1.10
 
 I tried /rtpengine_offer / in  kamailio.cfg, 
 but rtpengine seems to always use different ports to Tx/Rx the packets. 
 (Causing Freeswitch to auto adjust its ports, which mutes the audio
  channel one way).

Well, rtpengine actually uses the same port to send and receive traffic
towards a UA. However, it uses two different ports for the two parties,
which shouldn't be an issue, because Freeswitch will only see the one
port towards itself.

Andreas

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Re: [SR-Users] [rtpengine] No media from WebRTC UA

[Andreas Granig]

Hi,

There finally was some progress on the ice-lite issue in Firefox over
the last week or two (after the bug report got stuck for months), so I'd
expect Firefox nightly to properly work with ice-lice in a couple of
days or so.

Andreas

On 05/17/2014 02:30 AM, Alexey Rybalko wrote:
 Hi Richard!
 
 Just have tried an outgoing call to Chrome and Opera and it works fine.
 Thank you for the clarification regarding ICE! Dump files and rtp logs
 (Firefox, Chrome) I've sent to your email.
 
 My little investigation brings the following :
 
 - browsers (Firefox, Chrome  and Opera) don't use a=ice-lite in their SDP;
 
 - Chrome and Opera take the role of ICE-CONTROLLING and provide
 USE-CANDIDATE for the ICE-Lite peer (mediaproxy) for both offer and
 answer cases;
 
 - Firefox takes the role of ICE-CONTROLLING and provide USE-CANDIDATE in
 case of the offer while it takes the role of ICE-CONTROLLED during the
 answer;
 /means no media from WebRTC UA in the latter case/
 
 Some other remarks.
 
 During the call from Fire I saw a lot of SRTP output wanted, but no
 crypto suite was negotiated messages  from rtpengine. However DTLS is
 finally was established. Is that one more issue of Firefox?
 
 Looking in STUN section of the dump files I wonder why Chrome use more
 than 10 binding request (USE-CANDIDATE) for each candidate  while
 Mozilla does it just once.
 
 regards,
 Alexey
 
 
 
 2014-05-16 14:53 GMT+04:00 Richard Fuchs rfu...@sipwise.com
 mailto:rfu...@sipwise.com:
 
 
 There's nothing wrong with the SDP bodies that I can see. I recall that
 Firefox had or still has a problem with ICE role switching when ice-lite
 is offered. It never completes ICE negotiation (never sends an STUN
 packet with use candidate) and so never starts DTLS handshake.
 
 You can confirm that by doing a packet capture including the RTP ports
 and inspecting the STUN packets. Chrome shouldn't have that problem
 though, perhaps do another test run with it? You can send those capture
 files to me if you'd like me to have a look.
 
 
 
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[SR-Users] parallel forking and 30x

[Andreas Granig]

Hi,

What is the intended behaviour from a general point of view if you get a
1xx on one branch and a 30x on another branch for a parallel fork?

From my understanding, the 30x is a final reply, so it's kept in tm
until all other branches are finished, then it enters failure-route with
the highest error code (e.g. a 487 if you cancel the other branch).

What I actually want is intercepting the 30x and send another INVITE to
the Contact of the 30x in the failure route. This works fine if you only
have one branch, but it doesn't enter failure route for 30x if you have
another active branch (e.g. in ringing state). Would it be possible to
create a new branch directly from the reply route, or force to enter
failure route for a specific reply/branch?

Andreas

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Re: [SR-Users] parallel forking and 30x

[Andreas Granig]

Hi Daniel,

On 02/18/2014 10:38 AM, Daniel-Constantin Mierla wrote:
 starting with 4.1, you can have per branch failure routing block:
 
 http://www.kamailio.org/docs/modules/4.1.x/modules/tm.html#tm.f.t_on_branch_failure

Ha, we'll check if this is going to work for us, sounds good, thanks!
Just for clarification: the normal failure route will still be
executed after all branches are ended, right?

Andreas

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[SR-Users] broken ims modules links in online doc

[Andreas Granig]

Hi,

Seems like two ims module links (linked from
http://kamailio.org/docs/modules/4.1.x/) are broken:

http://kamailio.org/docs/modules/4.1.x/modules/ims_registrar_pcscf.html
http://kamailio.org/docs/modules/4.1.x/modules/ims_usrloc_scscf.html

The same for http://kamailio.org/docs/modules/devel/, as far as I can see.

Thanks for checking!

Andreas


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Re: [SR-Users] unregister user when kamailio looses TCP connection.

[Andreas Granig]

Hi Vitaliy,

On 09/24/2013 12:45 AM, Vitaliy Aleksandrov wrote:

The patch only handles the case where a tcp connection is directly
made to the registrar, as no event route is fired, right?

You are right. Current version works only when registrar accepts tcp
connections.
Anyway it's a good idea to call event_route[] when kamailio looses a tcp
connection to give user a chance to process it somehow.

On the one hand it's great when experienced user can achieve what he
wants by different ways depending on the situation, but on the other it
makes a mess for new kamailio users.


I think the implementation is fine for normal use cases where you just 
have one combined registrar/proxy, which is probably what new kamailio 
users are starting with.


The reason I'd love to see an event-route is to handle cases for scaled 
architectures, where you have load-balancers/edge-servers accepting 
tcp/tls connections from clients and forward them to a back-end farm of 
registrars/proxies e.g. via udp.


The implementation of that event-route would be completely independent 
of the current solution anyways, because it won't hook up with usrloc 
(on the edge proxies there is usually no usrloc loaded), rather than 
having the code somewhere in the core and just firing an event-route, 
where the system developer can do arbitrary actions (like creating a 
custom sip message to be sent to the back-end servers, which triggers an 
unregister there). Honestly I have no clue yet as of how to actually do 
that. Firing the event-route would be the rather easy part I guess, but 
do we have all the information needed to do an unregister on the 
back-end registrar? Probably the only info we have is the source 
ip/port/proto of the disconnected client, so we'd need some logic in the 
registrar/usrloc module on the back-end registrar to match a location 
entry with that info somehow.


I'm just throwing in some thoughs on that topic for further discussions, 
so no clear idea how to accomplish that from my side.


Andreas

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Re: [SR-Users] unregister user when kamailio looses TCP connection.

[Andreas Granig]

Hi,

On 09/13/2013 11:27 AM, Daniel-Constantin Mierla wrote:

thanks, patch was commited and pushed to remote repository.


The patch only handles the case where a tcp connection is directly made 
to the registrar, as no event route is fired, right?


Andreas

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Re: [SR-Users] mediaproxy-ng documentation available

[Andreas Granig]

On 07/07/2013 04:01 AM, Nick Khamis wrote:

Can we run it behind nat? I've really been eyeing that and desparate
to replace rtpproxy


Yes, check advertised address option.

Andreas

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Re: [SR-Users] mediaproxy-ng documentation available

[Andreas Granig]

Hi,

On 07/07/2013 11:19 AM, Raúl Alexis Betancor Santana wrote:

On Sat, Jul 06, 2013 at 10:01:28PM -0400, Nick Khamis wrote:

Can we run it behind nat? I've really been eyeing that and desparate
to replace rtpproxy


Think twice ... trying to run something designed to avoid problems
with NAT behing NAT? ... doesn't sound good at all ...


If it's behind a DNAT like on Amazon EC2, it could make sense.

Andreas

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[SR-Users] mediaproxy-ng documentation available

[Andreas Granig]

Hi,

There've been a couple of questions regarding mediaproxy-ng as an 
rtpproxy replacement on how to get it up and running with kamailio and 
what it does compared to rtpproxy, mediaproxy and iptrtpproxy.


I hope, the README.md at

https://github.com/sipwise/mediaproxy-ng

answers these questions and makes it easy for you to use it.

Feedback highly appreciated,
Andreas

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Re: [SR-Users] mediaproxy-ng documentation available

[Andreas Granig]

Hi,

On 07/05/2013 11:56 PM, Iwan Budi Kusnanto wrote:

I can see that SDES-SRTP is supported but DTLS-SRTP is not mentioned.
Is it mean that there is no support for DTLS-SRTP? Any plan to support it?


It's planned (like ZRTP), just no specific time-line yet.

Andreas

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Re: [SR-Users] Issue with RTP proxy....

[Andreas Granig]

Hi,

On 07/04/2013 09:00 AM, Khue Nguyen Minh wrote:

I am trying install module mediaproxy-ng, but, when I built module
kernel-module I got error:
make -C /lib/modules/2.6.32-279.22.1.el6.x86_64/build
M=/usr/local/src/mediaproxy_ngcp/mediaproxy-ng-master/kernel-module
O=/lib/modules/2.6.32-279.22.1.el6.x86_64/build modules
make: *** /lib/modules/2.6.32-279.22.1.el6.x86_64/build: No such file or
directory.  Stop.
make: *** [modules] Error 2

Modules daemon and iptables-extension built success.
How I can fix it.


You're missing the kernel headers. Having the kernel module is optional 
(as the daemon falls back to userspace-only mode if it fails to load the 
kernel module), but it's highly recommended for performance reasons.


Andreas

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Re: [SR-Users] Websocket transport and forward()

[Andreas Granig]

Hi,

Any feedback on that? Looks like the patch below does its job, can 
someone please confirm this is fine and actually as expected?


Andreas

On 06/27/2013 02:03 PM, Andreas Granig wrote:

Hi,

On 06/27/2013 12:54 PM, Andreas Granig wrote:

$du = $null;
$fs = $null;
forward();

What I get is this error message:

Jun 27 12:30:53 spce lb[12104]: ERROR: core [action.c:437]:
do_action(): ERROR: do action: forward: bad uri transport 5

Is there anything special I need to do? Does forward() maybe just not
understand WS transport?



Alright, I think I found it. It seems like there is an issue in action.c
where PROTO_WS and PROTO_WSS are not handled. This patch solves it for
me, however it's just a proof of concept as I don't know enough of the
core and WS implementation to understand the implications:

#+
diff --git a/action.c b/action.c
index e64cf81..58024a7 100644
--- a/action.c
+++ b/action.c
@@ -421,6 +421,8 @@ int do_action(struct run_act_ctx* h, struct action*
a, struct sip_msg* msg)
 /* no proto,
try to get it from the dns */
 break;
 case PROTO_UDP:
+   case PROTO_WS:
+   case PROTO_WSS:
  #ifdef USE_TCP
 case PROTO_TCP:
  #endif
#-

With that patch in place, I can set $fs to my local tcp socket, like

 $fs=tcp:192.168.51.133:5060;

and let $du point to the right destination, like

 $du=ws:192.168.51.1:1234;

and the request is forwarded correctly.

Any feedback on this?

Andreas

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Re: [SR-Users] Websocket transport and forward()

[Andreas Granig]

Hi Peter,

On 07/01/2013 12:38 PM, Peter Dunkley wrote:

The case PROTO_WS: should go within the #ifdef USE_TCP conditional
and the case PROTO_WSS: should go within the #ifdef USE_TLS conditional.


I'll clean it up and commit to master.

Thanks for confirmation,
Andreas

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[SR-Users] Websocket transport and forward()

[Andreas Granig]

Hi guys,

Maybe I'm missing something, but when I have a stateless lb with 
websockets enabled, and I don't have mhomed set (meaning that I need to 
set $fs manually), how would I set $du and $fs correctly, if $ru is for 
example this:


sip:bo87fffs@h7h3rerhg50k.invalid;transport=ws

I tried all kinds of variants and now starting with the simplest 
approach possible, doing this for sending out the request to the 
websocket client:


$du = $null;
$fs = $null;
forward();

What I get is this error message:

Jun 27 12:30:53 spce lb[12104]: ERROR: core [action.c:437]: 
do_action(): ERROR: do action: forward: bad uri transport 5


Is there anything special I need to do? Does forward() maybe just not 
understand WS transport?


I'm using kamailio 4.0.2.

Any help is greatly appreciated!
Andreas

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Re: [SR-Users] Websocket transport and forward()

[Andreas Granig]

Hi,

On 06/27/2013 12:54 PM, Andreas Granig wrote:

$du = $null;
$fs = $null;
forward();

What I get is this error message:

Jun 27 12:30:53 spce lb[12104]: ERROR: core [action.c:437]:
do_action(): ERROR: do action: forward: bad uri transport 5

Is there anything special I need to do? Does forward() maybe just not
understand WS transport?



Alright, I think I found it. It seems like there is an issue in action.c 
where PROTO_WS and PROTO_WSS are not handled. This patch solves it for 
me, however it's just a proof of concept as I don't know enough of the 
core and WS implementation to understand the implications:


#+
diff --git a/action.c b/action.c
index e64cf81..58024a7 100644
--- a/action.c
+++ b/action.c
@@ -421,6 +421,8 @@ int do_action(struct run_act_ctx* h, struct action* 
a, struct sip_msg* msg)
/* no proto, 
try to get it from the dns */

break;
case PROTO_UDP:
+   case PROTO_WS:
+   case PROTO_WSS:
 #ifdef USE_TCP
case PROTO_TCP:
 #endif
#-

With that patch in place, I can set $fs to my local tcp socket, like

$fs=tcp:192.168.51.133:5060;

and let $du point to the right destination, like

$du=ws:192.168.51.1:1234;

and the request is forwarded correctly.

Any feedback on this?

Andreas

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Re: [SR-Users] Kamailio: MI Command: 'trusted_dump' and 'trusted_reload'

[Andreas Granig]

Hi,

On 06/21/2013 10:59 AM, José Luis Millán wrote:

Hi,

I'm using 'allow_trusted()' from Permissions module without a problem in
a Kamailio 4.0 routing logic. It does just work, but the MI command
'trusted_dump' tells me that the module is not in use:


Seems like there is an issue with the rpc commands using kamcmd as well 
in kamailio 4:


kamcmd permissions.trustedDump
error: 500 - Reload failed. No trusted table

There *is* a trusted table in DB, and I'm using db_mode=0. Not sure why 
it tries a reload, maybe a copy/paste issue in the error string.


Andreas

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Re: [SR-Users] gruu and dead-lock in registrar module

[Andreas Granig]

Hi,

On 05/01/2013 10:46 AM, Daniel-Constantin Mierla wrote:

looked over the code and seems ok. The domain lock is set inside
ul.get_urecord_by_ruid(_d, ahash, inst, r, ptr).


Ok, good to know.


Are you doing other operations in config with usrloc/registrar rather
than save()/lookup()? Any mi/rpc commands? Any other modules bound to
usrloc (e.g., pua_usrloc)?


I'm regularly calling MI functions to get the number of records in 
usrloc. During the last deadlock scenario, I saw a kamctl fifo process 
querying usrloc hanging there since 2 days already, but I actually 
expected more of them. Didn't try to issue the MI command manually to 
see if that was responding, will do next time this happens.


Andreas

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Re: [SR-Users] [PATCH] Memory corruption using s.substr transformation

[Andreas Granig]

Hi,

We've seen this behaviour as well and worked around it using avp_subst 
with regex, as we didn't have the time yet to investigate further.


But basically I can confirm this issue.

Andreas

On 04/30/2013 01:31 PM, Martin Mikkelsen wrote:

On Tue, Apr 30, 2013 at 01:29:21PM +0200, Martin Mikkelsen wrote:

I am not sure what the best fix would be for that, but I have attached a
patch which copies the string in the TR_S_SUBST code to _tr_buffer and
returns that buffer instead like a lot of the other transformations in
that function does.

If this fix is correct some other transformation functions probably
needs to be corrected as well.


Forgot to attach the patch. Sorry.



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[SR-Users] gruu and dead-lock in registrar module

[Andreas Granig]

Hi,

We're hitting an issue in a deployment where all udp receivers are 
sitting in FUTEX_WAIT caused by save() - lock_udomain() and seem to 
have deadlocked themselves every couple of days.


Looking at the code, enable_gruu in registrar is active by default, and 
in lookup there is a code path


/* temp-gruu lookup */
res = ul.get_urecord_by_ruid(_d, ahash, inst, r, ptr);

but no lock_udomain is obtained. However, when the execution falls 
through to the done: marker, it does


ul.unlock_udomain(_d, aor);

without having called ul.lock_udomain first.

1.) Could someone please review this part? Looks a bit suspicious, 
although I don't know what implicitly happens in this case. If it were a 
semaphore and you decrease it to -1 by decrementing it without prior 
increment, it's essentially causing a dead-lock, but the current locking 
implementation might work completely different.


2.) Since I have no clue how gruu is supposed to work in detail, and 
since in our config we don't explicitly handle gruu (no lookup in 
loose-route, but gruu is enabled by default in registrar and we don't 
explicitly turned it off), I'm not even sure if we ever hit this code 
path. I only see that the ruid column in the location table is filled, 
but in order to get to this part, the ;gr flag needs to be set in the 
R-URI for a lookup(), which I don't know whether that happened somehow 
in some call flows (we only log $ru, which I don't think logs these 
parameters, right?).


Some input is highly appreciated!

Andreas

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Re: [SR-Users] gruu and dead-lock in registrar module

[Andreas Granig]

Hi Daniel,

On 04/30/2013 05:34 PM, Daniel-Constantin Mierla wrote:

what version are you playing with? To look in the right branch when
troubleshooting first time, then look at the others that might be
affected...


The affected version is latest 3.3 branch, but the same code is there in 
4.0 as well.


Andreas

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Re: [SR-Users] rfc: accounting records time details

[Andreas Granig]

Hi,

On 04/29/2013 01:42 PM, Alex Hermann wrote:

On Monday 29 April 2013 11:05:36 Daniel-Constantin Mierla wrote:

1) store seconds.miliseconds as double - there is a patch (which


Please do not use floating point respresentations for values that will be used
in accounting. Floating point is imprecise. As the time related columns will
most probably be used for billing, the values should be exact. In SQL this
means using the DECIMAL or NUMERIC column type.


Just for clarification, we're using DECIMAL(13,3) with the patch on the 
tracker, not DOUBLE. What's missing in this patch is the table 
definition, so it needs to be updated in any case.


Andreas

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Re: [SR-Users] rfc: accounting records time details

[Andreas Granig]

Hi,

On 04/29/2013 11:47 AM, Thilo Bangert wrote:

I'd just save one timestamp, ie TAI64 or as java does miliseconds since epoch,
in a single, new field. saving the timestamp in two fields seems messy.


Agreed.


miliseconds since epoch is probably preferable, since it can be converted to
human readable dates by the database server.


What I like about the DECIMAL approach is that it's (at least in MySQL) 
usable with from_unixtime functions, in case you need quick access to 
human readable format. Up until 5.1 it only shows seconds precision in 
that case, not sure about high resolution precision in 5.6 where 
timestamp seems to support microseconds.


Also not sure about compatibility with other DB engines.

Andreas

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Re: [SR-Users] Fwd: save accdb records with milisec in timestamp field

[Andreas Granig]

Hi Uri,


On 04/23/2013 10:45 AM, Uri Shacked wrote:

can anyone tell me if it is possible to save the time and date using
accdb (mysql) with milisec as well? the field on the DB is set as TIMESTAMP.


We've opened a ticket on the bug tracker some months ago, but it's still 
open at 
http://sip-router.org/tracker/index.php?do=detailstask_id=163project=1 
and we haven't followed up on it, however we're using this patch in 
production for months at all our deployments without any issues.


Andreas

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Re: [SR-Users] Fwd: save accdb records with milisec in timestamp field

[Andreas Granig]

Hi Daniel,

On 04/23/2013 08:51 PM, Daniel-Constantin Mierla wrote:


- the tm variable is declared set, but not used
- tz is also not used, gettimeofday() can take NULL as second parameter
and iirc, tz is obsolete
- I wonder if gettimeofday() can actually fail and return code should be
checked for error cases


I'll check with my colleagues regarding these questions and regarding a 
final patch, as it might have been modified in the meanwhile (the live 
version is at https://github.com/sipwise/kamailio).



- the names of the new column, respectively 'time_hires' sounds a bit
strange to me, does it have any special meaning the word 'hires'?


It's hires as in high resolution for the timestamp value.


Overall, wouldn't be better to keep the seconds and microseconds (as
returned in a timeval structure) in separate columns. That means keeping
the time column as it is and adding a new column for microsecs. Then
people can get the precision as they want, including only down to the
miliseconds if that is what they need.


The time column is still being filled as usual, so you can just stay 
with that format. However, the problem with this column is its date-time 
format, which causes lots of problems for calls going over daylight 
saving time changes and when it comes to handling different time zones 
in general, so time_hires uses a unix-timestamp with 3 digit precision, 
which makes this much easier to handle. And then again, if you want to 
use the time_hires column, you can still do from_unixtime in mysql to 
get a normal datetime format in seconds precision, no need to handle 
seconds and milliseconds separately.




I will add these notes on the tracker so it can be continued there for
development.


Alright, we'll continue from there and push this forward also on our 
side to get it to upstream.


Andreas

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Re: [SR-Users] mediaproxy-ng Tutorial

[Andreas Granig]

Hi,

On 04/10/2013 11:02 AM, Jon Bonilla (Manwe) wrote:

I think it is a bad idea to name the relay mediaproxy-ng and the
corresponding Kamailio module rtpproxy-ng.



Indeed


Well, the point is that it's just an enhancement of the rtpproxy 
module. In the past, the rtpproxy module was used to communicate with 
mediaproxy-ng (and can still be used that way), as mediaproxy-ng 
implements the rtpproxy protocol.


The idea behind rtpproxy-ng was to provide a reference implementation of 
an easily extensible control protocol (as we needed it for our ICE 
handling anyways), because there were some discussions and plans to 
rework rtpproxy towards such a kind of protocol as well (JSON was one of 
the formats, but we rather went with bencode as it's faster to parse and 
still somewhat human readable). Mind you we're still working on the 
protocol documentation.


Now the real question is whether it makes sense to extend rtpproxy to 
use this protocol as well, and in this case rtpproxy-ng as a module name 
makes perfect sense. It would probably be a good idea to merge it to 
rtpproxy module and just use that at some point (requiring to upgrade 
rtpproxy along with kamailio though, or control the protocol version via 
a module parameter).


If there are no intentions to develop rtpproxy further, it would make 
sense to name our module mediaproxy-ng instead of rtpproxy-ng. The 
module is still only in our repo and not pushed upstream exactly because 
of this kind of naming decision (besides the lack of documentation).


Andreas


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Re: [SR-Users] mediaproxy-ng Tutorial

[Andreas Granig]

Hi,

On 04/10/2013 02:54 PM, Juha Heinanen wrote:

i consider it a bad idea to call the new media proxy mediaproxy-ng,
because it gives impression that this is a new incarnation of ag projects
mediaproxy that has existed for years.


mediaproxy-ng also exists for years (since 2007, prior to the redesign 
of mediaproxy to become kernel based as well) and in fact was a 
replacement of the relay part of the mediaproxy project. It used and 
relied on the dispatcher-part and its control protocol of mediaproxy.


We additionally implemented support for the rtpproxy protocol in 2010 or 
so (but never worked on adapting it to any newer mediaproxy control 
protocols), and now it also supports a completely new control protocol 
implemented in rtpproxy-ng.


So basically it evolved from replacing a part of the mediaproxy project 
into a drop-in replacement of rtpproxy and since a few weeks could exist 
on its own with the rtpproxy-ng module. It just never changed its name.


Andreas


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Re: [SR-Users] loose_route received parameter question

[Andreas Granig]

Hi,

On 04/08/2013 11:50 AM, Juha Heinanen wrote:

i would expect that loose_route() would set $du to received parameter
value, but it does not, because i get to syslog:

Apr  8 12:44:54 wheezy1 /usr/sbin/sip-proxy[5723]: INFO: Trying to relay to du 
null or ru sip:47089231@192.98.102.10:41073;transport=tcp

how can i tell p2 to set $du to received uri?  do i need to add another
route header in p1 that contains received uri?


Do you have use_received=1 from path module enabled on p2? It registers 
a call-back to rr and sets $du accordingly if a received param is found 
in the first (own) Route header.


Andreas

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Re: [SR-Users] mediaproxy-ng Tutorial

[Andreas Granig]

Hi,

On 04/02/2013 05:14 PM, Dani Popa wrote:

as far as i know, mediaproxy by AG Projects it's also kernel based
forwarding, so i don't understand what is the reasons to use another
mediaproxy let's say mediaproxy-ng.


Well, mediaproxy-ng development has started like 6 years ago as a 
replacement for mediaproxy (which at that point was python based and 
didn't do any kernel stuff) in our commercial solutions. This is the 
reason why there is still the tcp option in mediaproxy-ng to support 
the old control protocol used by kamailio's mediaproxy module. Around 
2-3 years ago we made it open source along with the sip:provider CE 
appliance.


The mediaproxy project has since evolved also to a kernel based project, 
and afaik the control protocol has changed as well, so both projects are 
not compatible anymore. I can't tell for sure the status of mediaproxy 
as we didn't use it since years. They are more opensips focused, but it 
probably works with kamailio as well (don't know though).


Since we switched to rtpproxy control protocol, it's a drop-in 
replacement for the rtpproxy project, which is pure user-space. There is 
some other rtpproxy control protocol compatible media replay 
(ipt_rtpproxy?) which I haven't used myself either.


So all in all, it's just about diversity, choose whatever suits you 
best. The only thing I can say for that matter is that Sipwise actively 
implements new features on top of mediaproxy-ng (with the new 
rtpproxy-ng kamailio module using a dictionary-based control protocol, 
which makes extending the protocol much easier in the future, and which 
passes back and forth the whole SDP body instead of just passing in some 
parameters and getting back out a new IP and port, which allows for 
advanced stuff like ICE manipulation, and in the future also transcoding 
etc).


Andreas

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[SR-Users] base64 transformation

[Andreas Granig]

Hi guys

Maybe I missed something, but looks like there is no base64 
encode/decode transformation in PV, right?
I think we're going to add one, as it might be handy to do such 
transformations directly from within the config file.


One case where this could be used is that some UAs are picky in regards 
to URIs as URI params, e.g.


P-Foo: sip:al...@example.org;x='sip:1.2.3.4:1234'

which could be changed into

P-Foo: sip:al...@example.org;x=c2lwOjEuMi4zLjQ6MTIzNAo=

which might be understood more easily (although it's still just a 
work-around for broken UAs, but that's how life is).


Any objections against such a transformation?

Andreas

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Re: [SR-Users] Kamailio strange behavior : bad Via, bad port

[Andreas Granig]

Hi,

On 03/18/2013 08:33 AM, Khoa Pham wrote:

I heard some people said that it is the ALG of some routers that modify
the SIP message, hence cause the errors


It's just guessing and consulting the magic crystal ball. Post your SIP 
message which kamailio complains about as it is received on your SIP 
server, so people can tell you what's wrong with it.


Andreas

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Re: [SR-Users] Weird out-of-memory message on start/stop

[Andreas Granig]

Hi Daniel,

On 02/21/2013 09:38 AM, Daniel-Constantin Mierla wrote:

My curiosity now, what means the syslog message:

... t of memory [6720]

I guess the last is pid. Is it like just last part of OUt of memory?
Never had this message so far, iirc.


What I found through debugging is that this message should actually mean 
out of memory, and it's yielded by libc, I guess during mmap and/or 
locking the memory (not exactly sure what kamailio does in that regards).


Andreas

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Re: [SR-Users] Weird out-of-memory message on start/stop

[Andreas Granig]

Hi,

On 02/21/2013 07:50 PM, Konstantin M. wrote:

Try to look at https://bugzilla.redhat.com/show_bug.cgi?id=803827  and
http://forums.cpanel.net/f34/named-wont-start-284821.html#post1191151


As noted above, I've found the issue already, which was caused by 
setting the lock-mem ulimit (the -l switch) too low.


Andreas

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Re: [SR-Users] SBC

[Andreas Granig]

Hi,

On 02/20/2013 04:24 PM, Radek Wal wrote:

SEMS als SBC application.


You might want to check 
http://www.sipwise.com/news/technical/byov-system-spce-as-sbc/ which 
uses kamailio and sems to do exactly that.


Andreas

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Re: [SR-Users] Nonce secret and expiry value

[Andreas Granig]

Thanks for clarification, Daniel!

Andreas

On 02/19/2013 09:18 AM, Daniel-Constantin Mierla wrote:

Hello,

On 2/18/13 4:40 PM, Andreas Granig wrote:

Hi guys,

Nonces have a default expiry value of 300 seconds. Does this still
apply if the secret parameter is set? So can you use both secret
and nonce_expire value together, taking for granted that the time is
in sync over multiple proxies?

yes, both work together and secret has to be set if you want to sync
across multiple servers. If secret is not set, then its value is
randomly generated at startup.

Cheers,
Daniel



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[SR-Users] Weird out-of-memory message on start/stop

[Andreas Granig]

Hi,

Maybe some of you have encountered this funny error message when 
starting and even killing kamailio on a Debian 6 with 2.6.x and 3.x 
kernels and 2.11.3-4:


#-
# /usr/sbin/kamailio -f /etc/kamailio/proxy/kamailio.cfg -P 
/var/run/kamailio/kamailio.proxy.pid -m 4 -M 16 -u kamailio -g kamailio

loading modules under /usr/lib/kamailio/modules_k:/usr/lib/kamailio/modules
Listening on
 udp: 127.0.0.1:5062
 tcp: 127.0.0.1:5062
Aliases:


Message from syslogd@sp2 at Feb 19 11:15:42 ...
 t of memory [6720]

# killall kamailio

Message from syslogd@sp2 at Feb 19 11:17:51 ...
 t of memory [6720]

#-

The PID reported here is the kamailio attendant process. It seems to be 
somehow related to the shared memory. The interesting part about that is 
that kamailio really starts just fine, and there is plenty (5GB) of 
free memory available. It doesn't seem to influence the behaviour of 
kamailio at all as it's working perfectly fine, so for now it's just 
annoying, but I'm really curious


I've tried to fiddle with mem_join, but it doesn't matter whether it's 
on or off.


Kamailio version is this:

# kamailio -V
version: kamailio 3.3.2 (x86_64/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, 
DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, 
F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, 
USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 
MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB

poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: unknown
compiled on 19:09:17 Jan 21 2013 with gcc 4.4.5

Has anyone seen this too? What's the cause of that?

Andreas

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Re: [SR-Users] Weird out-of-memory message on start/stop

[Andreas Granig]

On 02/19/2013 11:24 AM, Andreas Granig wrote:

Maybe some of you have encountered this funny error message when
starting and even killing kamailio on a Debian 6 with 2.6.x and 3.x
kernels and 2.11.3-4:


2.11.3-4 is the libc version, btw.

Andreas

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Re: [SR-Users] Weird out-of-memory message on start/stop

[Andreas Granig]

Hi,

On 02/19/2013 11:26 AM, Andreas Granig wrote:

On 02/19/2013 11:24 AM, Andreas Granig wrote:

Maybe some of you have encountered this funny error message when
starting and even killing kamailio on a Debian 6 with 2.6.x and 3.x
kernels and 2.11.3-4:


2.11.3-4 is the libc version, btw.


Oh. Setting ulimit -l unlimited does the trick.

Andreas

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[SR-Users] Nonce secret and expiry value

[Andreas Granig]

Hi guys,

Nonces have a default expiry value of 300 seconds. Does this still apply 
if the secret parameter is set? So can you use both secret and 
nonce_expire value together, taking for granted that the time is in 
sync over multiple proxies?


Andreas

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[SR-Users] uac_replace_from recovery with modified header

[Andreas Granig]

Hi,

This is a question for kamailio 3.0.x and I'm still trying to reproduce 
this issue somehow for latest stable, but maybe it was a known issue 
anyways which got fixed in newer versions, so I don't have to dig deeper 
into this:


There is an INVITE A - kamailio - B, and kamailio does 
uac_replace_from() to replace the display- and user-part (leaving the 
domain-part untouched).


When there's a BYE B - kamailio - A, then From/To are switched, so 
the replaced From of the INVITE becomes the replaced To of the BYE. 
Kamailio gets that one right and tries to recover (in auto-mode) the To.


However, in this BYE, B changes the domain-part of the To, which should 
be valid, as elements are only supposed to inspect the tags of From/To, 
right? The problem though is that when recovering the To, kamailio 
inserts garbage into the domain part. The vsf-Parameter in the Route of 
the BYE is definitely ok though, so that doesn't seem to be the issue.


Here's an example:

In the INVITE,
From: ab12345sip:ab12345@192.168.0.12;user=phone;tag=948a3340
gets rewritten to
From: 12345sip:12345@192.168.0.12;user=phone;tag=948a3340

And in the BYE,

To: 12345sip:12345@1.2.3.4;tag=948a3340
gets rewritten to
To: 12345sip:ab12345@-%:68*0.0%12;user=phone;tag=948a3340

Note that the received To in the BYE has a different host-part than what 
got sent out in the From of the INVITE, and when kamailio recovers the 
To, it doesn't recover the Display-part (not sure though if this is 
intended behaviour anyway), but it only party recovers the domain-part. 
You can still see some artefacts of the original domain (like the 68 
from the second 168 octet, and the 12 from the last octet).


Has anyone encountered this issue before?

Andreas

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Re: [SR-Users] uac_replace_from recovery with modified header

[Andreas Granig]

Hi Daniel,

On 01/24/2013 03:55 PM, Daniel-Constantin Mierla wrote:

The latest version has support for using dialog to store the From/To
values, so no masking within rr param. It should work fine in your case,
I guess.

For earlier version, you can achieve more or less the same by storing
from/to values in htable, indexed by call-id.


Thanks for the explanation.

This issue actually happens on our very first SPCE version (2.1), which 
just runs one kamailio as proxy without any load-balancer and sbc in 
between the proxy and the called party.


Since 2.2 we use sems as sbc between proxy and lb on the way out to the 
called party b, so in that case if B sends something odd in the To, 
it'll be discarded and replaced by the To (or From, depending on the 
direction) which had been passed from the proxy to the sbc in the 
initial INVITE.


So that's for example a very good scenario where it pays off to have an 
SBC in your system :)


Andreas

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Re: [SR-Users] uac_replace_from recovery with modified header

[Andreas Granig]

Hi,

On 01/24/2013 04:49 PM, Daniel-Constantin Mierla wrote:

there is no need for an sbc and break the call in two legs and drop my
cool extensions I have in my softphone.


What cool extensions would that be? I guess the sems guys would be happy 
to fix it, if it breaks something unexpectedly.



The comparison is more like: it's better to have an airplane instead of
a car to drive on highway because can be done without wearing safety
belt, which is required only for take off, landing and turbulence :-).


An SBC is rather like an airbag. If done right, you don't even recognize 
it and in case of an issue it might save your ass. If done wrong, it 
will blow up in your face while driving 100mph on the highway.


Andreas

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Re: [SR-Users] Send reply with Reason header

[Andreas Granig]

Hi,

On 12/19/2012 05:49 PM, Mino Haluz wrote:

I know how to use sl_send_reply, but I would like to add Reason:
header to the reply generated. Is there any command or variable for that?


You can use append_to_reply() from textops module: 
http://kamailio.org/docs/modules/1.3.x/textops.html#AEN276


Andreas

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Re: [SR-Users] Changing METHOD

[Andreas Granig]

Hi Olle,

On 12/19/2012 02:43 PM, Olle E. Johansson wrote:

Yeah, I already have Kamailio sending all kinds of crazy stuff while
testing a new platform,
using the UAC module. But I wanted to be lazy and change on the fly...

Evil stuff happening here. Kamailio is a good test-tool.

Now I fail to change Contact: headers. The docs for textops say that
remove_hf can remove
contact and Remove_hf(Contact) returns true - but the old contact is
still there!


You could try using msg_apply_changes() from 
http://kamailio.org/docs/modules/stable/modules/textopsx.html#textopsx.msg_apply_changes 
after remove_hf and before actually building your new request with 
$uac_req, but this is really dangerous.


One thing I know for sure is that you better not call record_route() 
anywhere before msg_apply_changes(), as record_route() only adds a hint 
to the message and the value is filled in once the sending socket is 
known (manually setting $fs doesn't seem to help). If you call 
msg_apply_changes() after record_route(), you'll get errors and will end 
up with a broken Record-Route header.


Hope this helps,
Andreas

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Re: [SR-Users] Send reply with Reason header

[Andreas Granig]

Hi Daniel,

On 12/19/2012 07:24 PM, Daniel-Constantin Mierla wrote:

You can use append_to_reply() from textops module:
http://kamailio.org/docs/modules/1.3.x/textops.html#AEN276

:-) I couldn't go over not to remark your good spot on the docs of the
7th older release by now - the 1.3.x - which is the same content for
this functions as for latest version (3.3.x).


An unintentional tribute to my favourite version :)

Cheers,
Andreas

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Re: [SR-Users] variable in rewritehost ?`

[Andreas Granig]

Hi,

This might sound like a silly question, but why don't you use lookup() 
instead? You can pass a PV also instead of just using $ru.


Andreas

On 12/17/2012 04:40 PM, Ali Jawad wrote:

Seems the logic I used is flawed, problem is not all of the sip clients
register their domain in the location table, alot of them have NULL as
domain, so I need to rely on the socket column in location table to
determine where to send the incoming call. Problem is I need to extract
only the IP from udp:xx.xx.xx.xx:5060 , any idea on how to achieve that ?
Regards

On Mon, Dec 17, 2012 at 4:13 PM, Alex Balashov
abalas...@evaristesys.com mailto:abalas...@evaristesys.com wrote:

Your reference to the row, column lacks $dbr() PV encapsulation.

Ali Jawad ali.ja...@splendor.net mailto:ali.ja...@splendor.net
wrote:

 Thanks Alex, now it works if I do set the $ru manually, however if
I do
 set
 it dynamically it still gives destination unresolvable, I do fetch a
 SQL
 statement and row 1,1 is hostname
 
 
 while($var(i)$dbr(ra=rows))
 
 {
 
 $var(j) = 0;
 
 while($var(j)$dbr(ra=cols))
 
 {
 
 xlog([$var(i),$var(j)] =
 $dbr(ra=[$var(i),$var(j)])\n);
 
 $var(j) = $var(j) + 1;
 
 }
 
 $var(i) = $var(i) + 1;
 
 }
 
 }
 
 
 
 $ru = sip:support1@ra=[$var(1),$var(1)]:5060;
 
 
 On Mon, Dec 17, 2012 at 2:22 PM, Alex Balashov
 abalas...@evaristesys.com mailto:abalas...@evaristesys.comwrote:
 
  $ru

--
Sent from my mobile, and thus lacking in the refinement one might
expect from a fully-fledged keyboard.

Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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--
*Ali Jawad
*
*Information Systems Manager
CISSP - PMP - ITIL V3 - RHCE - VCP - C|EH - CCNA - MCSA
*
*Splendor Telecom (www.splendor.net http://www.splendor.net/)
Beirut, Lebanon
Phone: +9611373725/ext 116
FAX: +9611375554

*



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Re: [SR-Users] NAPTR, SRV and sips vs. transport=tls

[Andreas Granig]

Hi Klaus,

On 12/03/2012 10:15 AM, Klaus Darilion wrote:
 The request URI should look like the one which the user enters. E.g. if
 user enters sip:12...@example.com then the request URI should be
 sip:12...@example.com - regardless of the transport protocol chosen by
 the transport layer.
 
 Thus, if NAPTR records tell you to use SIP over TLS, then use SIP over
 TLS but do not change the request URI.

So how should the NAPTR record look like if you want to use TLS with a
SIP URI? Would it still be SIPS+D2T, or could you use something like
SIP+D2T along with a replacement part _sip._tcp.example.com;transport=tls?

Andreas



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Re: [SR-Users] NAPTR, SRV and sips vs. transport=tls

[Andreas Granig]

Hi Klaus and Ole,

On 12/03/2012 11:43 AM, Klaus Darilion wrote:
 Just use:
 @ IN NAPTR 50   50  s  SIPS+D2T   _sips._tcp.example.com.
 
 I would interpret it as:
 
SIPS+D2T
  | \
  |  \
 secure + TCP -- TLS

Ok, this seems like a valid approach. I'll investigate further how
different clients are actually interpreting it and reacting on this.

Thanks,
Andreas



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[SR-Users] NAPTR, SRV and sips vs. transport=tls

[Andreas Granig]

Hi,

Hope to get some guidance here over the usage of sips and sip plus
transport=tls when following RFC3263.

The RFC3263 says that a NATPR record could return something like this
for a query like host -t NAPTR example.com:

   ;  order pref flags service  regexp  replacement
  IN NAPTR 50   50  s  SIPS+D2T   _sips._tcp.example.com.
  IN NAPTR 90   50  s  SIP+D2T_sip._tcp.example.com
  IN NAPTR 100  50  s  SIP+D2U_sip._udp.example.com.

This means that the client should use sips if possible when contacting
example.com.

Concluding from that, I suppose it should perform an SRV lookup host -t
SRV _sips._tcp.example.com, and the result would be:

   ;;  Priority Weight Port   Target
   IN SRV  01  5061   server1.example.com
   IN SRV  02  5061   server2.example.com

What I'm curious about is how requests towards one of these servers
should look like, and how they are being handled by kamailio.

A lot of clients and servers are sending
sip:u...@example.com;transport=tls in request URIs and Contact headers
and Record-Route headers, and you can check with
uri_param(transport,tls) which transport socket to use. This is
pretty useful as you can determine hop-by-hop whether or not to use TLS.
This approach has been obsoleted by RFC3261 though, and there doesn't
seem to be a mechanism in RFC3263 to indicate use schema sip, but use
transport=tls.

On encounter of a NAPTR record like the one above, how does kamailio
act? Does it set a sips schema for the next hop?

And what's the general take on this sips schema? As far as I
understand RFC3261, it means that if a client sends a request to a
sips-URI, the request is sent to the domain via TLS, and from there the
request is sent securely to the callee, but with security mechanisms
that depend on the policy of the domain of the callee. (RFC3261,
Chapter 4). What does this really mean in practice? Are you allowed to
rewrite the schema to sip and pass it on for example via UDP to the
callee if the callee didn't indicate transport=tls (deprecated anyways)
or sips: in the Contact of the registration? Or should you keep sips
as schema, but still send it via UDP, because you know based on local
policy or based on client registration that the next hop is not
supporting TLS? How would widespread clients react when getting a call
to a sips URI, especially if they receive it via UDP?

Looking forward hearing your input on that,
Andreas






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Re: [SR-Users] NAPTR, SRV and sips vs. transport=tls

[Andreas Granig]

Hi Juha,

My impression is that transport=tls is something like a widely used
consensus on how to securely communicate in SIP. However, is there a way
how support for that is communicated towards a client, like SIPS+D2T was
supposed to be the way to indicate sips support in NAPTR?

I'm using TLS for some months now with a couple of clients, and I'm
starting to wonder why we're all still using a mechanism deprecated in
RFC3261, whereas we're usually quite sensitive to shortcomings and
mistakes in RFC3261 and are not shy cranking out new SIP related RFCs
like there's no tomorrow...

It didn't bother me much until today where I realized there is no way to
properly indicate to a client that my server supports TLS. Or is there?

Andreas


On 12/01/2012 12:29 AM, Juha Heinanen wrote:
 Andreas Granig writes:
 
 A lot of clients and servers are sending
 sip:u...@example.com;transport=tls in request URIs and Contact headers
 and Record-Route headers, and you can check with
 uri_param(transport,tls) which transport socket to use. This is
 pretty useful as you can determine hop-by-hop whether or not to use TLS.
 This approach has been obsoleted by RFC3261 though, and there doesn't
 seem to be a mechanism in RFC3263 to indicate use schema sip, but use
 transport=tls.
 
 when rfc3261 was specified, the whole sips thing was put there at the
 last minute.  no-one had any practical experience of it.  i guess the
 requirement to have something like https came from ietf area directors.
 the result is a useless mess.
 
 And what's the general take on this sips schema? As far as I
 understand RFC3261, it means that if a client sends a request to a
 sips-URI, the request is sent to the domain via TLS, and from there the
 request is sent securely to the callee, but with security mechanisms
 that depend on the policy of the domain of the callee. (RFC3261,
 Chapter 4). What does this really mean in practice? Are you allowed to
 rewrite the schema to sip and pass it on for example via UDP to the
 callee if the callee didn't indicate transport=tls (deprecated anyways)
 or sips: in the Contact of the registration?
 
 in that case, the proxy has to drop the request unless the last hop is
 secure, i.e., uses a vpn, barb wire, or something.
 
 Or should you keep sips
 as schema, but still send it via UDP, because you know based on local
 policy or based on client registration that the next hop is not
 supporting TLS? How would widespread clients react when getting a call
 to a sips URI, especially if they receive it via UDP?
 
 you cannot use sips with udp transport.
 
 -- juha
 
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Re: [SR-Users] ul.dump Problem

[Andreas Granig]

Hi,

On 11/26/2012 06:25 PM, Spencer Thomason wrote:
 I'm attempting to dump the contents of the usrloc cache and I'm receiving the 
 following error:
 sercmd ul.dump
 error: 500 - Internal error creating aor struct
 sercmd mi ul_dump
 error: 500 - Internal server error processing 's': buffer too small 
 (overflow) (-2)
 
 It seems to work fine when there are only a small number of registrations 
 (about 20 or so) but then fails when the number increases.  It there a 
 setting I need to increase somewhere?

Which version are you running? This has been fixed quite recently.

Andreas



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Re: [SR-Users] ul.dump Problem

[Andreas Granig]

On 11/26/2012 06:49 PM, Spencer Thomason wrote:
 I'm running 3.3.2 @ commit bab07e07858464d50d310bbb52431a0b171ee771

Should be fixed here:
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=3d195f2675569954a1f74128508db07cbc604ed9

Andreas



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[SR-Users] Sync nonce between various servers

[Andreas Granig]

Hi,

There are lots of parameters controlling the creation of nonce values on
a server, and I'm curious if there is a way to kind of sync them
between servers.

The use case would be to have a UA send for example its registration to
Proxy1. Proxy1 would challenge it, UA will send the registration again,
this time with credentials. Proxy1 would look up the user based on
$au/$ar in the subscriber table, and if it's not found, will look up the
responsible proxy from another table (with key being $au@$ar), forward
it to Proxy2, which then would be able authenticate the user.

The reason for this is that the auth credentials are unique across all
servers and reliably identify a user, whereas for example From could be
something else (e.g. in case of an IP-PBX sending a CLI in the
From-userpart).

Challenging the user on the second proxy again would theoretically be
possible, but if the UA gets a 401 twice (once from Proxy1, once from
Proxy2), it'll most likely pop up a password form for soft-clients, so I
want to avoid that.

Any ideas how to accomplish that?

Andreas





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Re: [SR-Users] Sync nonce between various servers

[Andreas Granig]

Hi David,

On 11/19/2012 02:54 PM, David J wrote:
 Is the database shared? If so maybe when they authenticate add a secure
 token to the header that the second proxy can use for auth?

No, the DBs are explicitely NOT shared in this scenario.

 Just a suggestion not sure if its the answer your looking for or perhaps
 I didn't understand the scenario well enough.

Let me try to put the scenario in different words:

If a request from a subscriber hits a server, and it doesn't contain an
Authorization header, then the server would just challenge the request.
This doesn't require any subscriber information on this server, so it
shouldn't matter whether this subscriber exists on this server or not.

When the request comes in again, this time with an Authorization header,
the server can use the username and realm of this header to check
whether the subscriber is local or not. If it's local, it would just try
to authenticate it as usual, and if it's not, it can look up the correct
server using this auth username/realm and forward the request to the
responsible server.

Now this second server would receive a request, which already contains
an authorization header, but it won't be able to authenticate it if the
nonce is not in sync between server1 and server2.

So this leads to the question whether it's possible to sync the nonces
in a way that server1 challenges a request, and a different server would
be able to authenticate the subsequent request holding the
challenge-response.

Andreas



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Re: [SR-Users] Sync nonce between various servers

[Andreas Granig]

Thanks Olle and Carsten,

On 11/19/2012 03:27 PM, Carsten Bock wrote:
 short question:
 Why don't you use a shared secret to create a nonce value?
 
 http://kamailio.org/docs/modules/devel/modules/auth.html#auth.secret

I've noticed this secret parameter, but the documentation is a bit
brief on the exact meaning of it, thus my question on the list.

If this setting is really doing what we all think it is doing, then
that'll be great! :)

I'll just try it out...

Andreas



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Re: [SR-Users] Sync nonce between various servers

[Andreas Granig]

Hi Carsten,

On 11/19/2012 03:38 PM, Carsten Bock wrote:
 I am using it with 3.2 in production for exactly that purpose 

Seems to work like charm, thanks!

Andreas



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Re: [SR-Users] Manually add Via Header

[Andreas Granig]

Hi,

Sorry for hijacking this thread, but I've a similar but different
question which bugs me since a while :)

Is there a way in kamailio to statelessy forward a request without
putting its own Via header into the message? Consider an example where a
stateless load-balancer sends a request to a proxy A, which again
statelessy forwards it to another proxy B, while it's not interested in
staying in the path even for replies. It's not necessary either because
the load-balancer, which would receive the replies from B then, can pass
them along according to the Via path, so no tm or something is involved.

This might break some RFC, but it would remove proxy A completely out of
the path once the request got forwarded. In theory this seems possible
if it's ensured that both the load-balancer and proxy A act strictly
stateless, no?

Andreas

On 10/25/2012 12:03 PM, Daniel-Constantin Mierla wrote:
 You better use t_reply(487, Cancelled) in a failure_route.
 
 Adding a proper Via header might be a tricky task.
 
 Cheers,
 Daniel
 
 On 10/25/12 11:43 AM, Vassilis Radis wrote:
 I have a situation where a far end SIP provider doesn't behave
 properly when sending 487 replies. The scenario is this:

 I have a registered user calling into my kamailio which ,using lcr
 module, routes the call to a SIP provider. When the caller Cancels the
 call, my kamailio forwards the cancel message to the provider (along
 with a 200 Cancelling message to the caller). The problem is with the
 487 sent by the provider to my kamailio: It does not contain all the
 two Via headers. It only contains the via header that names my
 kamailio. So when kamailio gets the 487, it removes its Via header and
 sends the 487 back to the caller, which now doesnt find any via header. 

 Note that the far end provider, correctly sends all the Via Headers in
 other replies (183 etc). 

 So now I am trying to intercept the 487 from this specific provider
 in the onreply route, and patch it with the header that the provider
 should have included. How can I do it? The problem is twofold:
 1.I need to detect which header I should manually add (maybe store the
 Via Header set from the initial invites or the 183 replies, and get it
 from there? .I dont know how to do that)
 2.I need to add it. (How can I manually add a Via header? .This seems
 easier

 Can anyone help me, maybe with another solution? 


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 -- 
 Daniel-Constantin Mierla - http://www.asipto.com
 http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
 Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
 Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - 
 http://asipto.com/u/katu
 
 
 
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Re: [SR-Users] Manually add Via Header

[Andreas Granig]

Hi,

On 10/25/2012 02:51 PM, Olle E. Johansson wrote:
 Is there a way in kamailio to statelessy forward a request without
 putting its own Via header into the message? Consider an example where a
 stateless load-balancer sends a request to a proxy A, which again
 statelessy forwards it to another proxy B, while it's not interested in
 staying in the path even for replies. It's not necessary either because
 the load-balancer, which would receive the replies from B then, can pass
 them along according to the Via path, so no tm or something is involved.

 This might break some RFC, but it would remove proxy A completely out of
 the path once the request got forwarded. In theory this seems possible
 if it's ensured that both the load-balancer and proxy A act strictly
 stateless, no?
 From the cookbook on the wiki:
 
 /O
 
 send
 
 Send the original SIP message to a specific destination in stateless mode. No 
 changes are applied to received message, no Via header is added. Host can be 
 an IP or hostname. Used protocol: UDP
 
 Parameter is mandatory and has string format.
 
 Example of usage:
 
  send(10.10.10.10:5070);

Oh, thanks for the hint! I was always thinking about using forward().
Will double-check send() again!

Andreas



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Re: [SR-Users] udp transport in rtpproxy sockets

[Andreas Granig]

Hi Juha,

On 10/22/2012 09:50 AM, Juha Heinanen wrote:
 Definition of socket(s) used to connect to (a set) RTPProxy. It may
 specify a UNIX socket or an IPv4/IPv6 UDP socket.
 
 if that is the case, what is the justification for it, i.e, why is tcp
 (and better yet tls) transport not supported?

I fully agree that TCP (as a first step) should be a good improvement.
We've already briefly talked about redesigning the overall rtpproxy
control protocol some time ago, but we haven't had the time yet to start
with it, however we've it on our list for December.

The idea is to switch to TCP, and to pass on the whole SDP body to the
rtp proxy, along with a more flexible way to pass parameters back and
forth (e.g. passing back RTP stats in the response of a delete-message).
The protocol will be something like json or bencode. Letting the rtp
proxy analyze the full SDP body and pass back a full SDP body in return
will allow us to do advanced stuff like RTP/AVP to RTP/SAVPF conversion
and ICE handling, beside SRTP transcoding etc.

I'm aware that this goes against the ideal world of end-to-end media
negotiation, but let's be pragmatic here. We know it's not realistic to
have all SIP clients updated and play nice in a reasonable time frame,
and the only other alternative is putting an Asterisk (trunk version) or
a similar application server in between, which is not necessary.

Most likely we're going to start with forking the rtpproxy module in a
separate feature branch, do a proof-of-concept implementation and
preferably merge it back to trunk rtpproxy module once all 3 media
proxies (rtpproxy, iptrtpproxy, mediaproxy-ng) are supporting this protocol.

Andreas



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Re: [SR-Users] mediaproxy-ng fails to parse start_recording() command

[Andreas Granig]

Hi Juha,

On 10/20/2012 07:17 PM, Juha Heinanen wrote:
 i tried what happens when i call start_recording() after calling
 rtpproxy_manage() and got this kind of error to syslog:
 
 Oct 20 19:57:43 siika mediaproxy-ng[8423]: Failed to properly parse UDP 
 command line '8554_13 R ngsgznnwgzpu...@siika.tutpro.com mgpwe' from 
 127.0.0.1:51497, using fallback RE
 
 why the error? it is so that mediaproxy-ng does not support recording?

Right, recording is not yet supported.

Andreas



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[SR-Users] Syntax of xavp

[Andreas Granig]

Hi,

I'm playing around with xavp, but there are some things I can't wrap my
head around.

What basically works is this:

$xavp(a=foo) = 'foo';
$xavp(a[0]=bar) = 'bar';

Getting the value like this:

xlog(L_INFO, a-foo='$xavp(a=foo)' and a-bar='$xavp(a=bar)');
as well as
xlog(L_INFO, a-foo='$xavp(a[0]=foo)' and a-bar='$xavp(a[0]=bar)');

... works, because when omitting the index, it assumes [0]. I really
like to avoid such implicit assumptions though, to make the config
really clear, so I prefer the second approach.

What doesn't work for me then is explicit assignment of the very first
value, like this:

$xavp(a[0]=foo) = 'foo';
$xavp(a[0]=bar) = 'bar';

If I do this, the a xavp is never created. Wouldn't it be good
practice to create the xavp on first assignment? That way, I don't have
to take care in the config file whether the xavp is used for the first
time, where I have to omit the index.

What I'm also wondering is whether it's possible to directly access
nested xavps, like this:

$xavp(a=foo) = 'afoo';
$xavp(b=foo) = 'bfoo';
$xavp(c=a) = $xavp(a);
$xavp(c=b) = $xavp(b);
xlog(L_INFO, a-foo='$xavp(c=a[0]=foo)');
or
xlog(L_INFO, a-foo='$xavp(c=a=foo)');

Both ways give me a-foo='xavp:0x7f0d387fe178'. Is this even
intended? Would be really cool if that's possible!

Thanks,
Andreas



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Re: [SR-Users] Kamailio Presence module

[Andreas Granig]

On 09/17/2012 10:36 AM, Gary Shergill wrote:
 root@opensep:~# service kamailio restart
 Not starting Kamailio: invalid configuration file!
 -e
  0(7892) : core [cfg.y:3591]: parse error in config file
 /etc/kamailio/kamailio.cfg, line 392, column 32-36: syntax error
  0(7892) : core [cfg.y:3591]: parse error in config file
 /etc/kamailio/kamailio.cfg, line 392, column 32-36: Invalid arguments
  0(7892) : core [cfg.y:3594]: parse error in config file
 /etc/kamailio/kamailio.cfg, line 392, column 37:
 ERROR: bad config file (3 errors)

Check what's in line 392, it should give you a hint what's wrong.

 Anyone knows what is wrong please? Hoping to get presence and IM
 working on the Kamailio server (enabling this presence module enables
 IM, correct?).

I'm also not familiar with the default kamailio config, but presence
itself has nothing to do with IM per se. There are two ways doing IM in
SIP: one is page-mode using MESSAGE requests, which should take pretty
much the same route as INVITE (that is: authorization of caller,
normalization of callee, lookup in location table, then t_relay to
callee); the other one is MSRP, which is actually an INVITE with a
special payload. If you use sylkserver, then you can just forward the
INVITE to sylkserver, which has an MSRP relay integrated, otherwise you
can use http://kamailio.org/docs/modules/3.3.x/modules/msrp.html to
handle MSRP directly within kamailio without a 3rd party server.

Andreas



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Re: [SR-Users] [OT] the role of SBCs

[Andreas Granig]

On 08/31/2012 11:10 AM, Olle E. Johansson wrote:
 Sbc also ofer a propritary way of failover for itself
 If node1 die node2 will replace it
 
 That's been done both with Asterisk and Kamailio (and propably SEMS) for a 
 very long time.

For SEMS it's a commercial module, same goes for the Sipwise mediaproxy
for seemless RTP stream fail-over. That's still where open source people
earn their money :)

Andreas



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Re: [SR-Users] [OT] the role of SBCs

[Andreas Granig]

Hi Rupert,

The interesting thing is that the arguments you brought up are exactly
the same in every big corporation we've worked with. They're perfectly
valid, also taking into account the decision making process, which
department takes responsibility for what, and it's also of course
politically and policy driven.

Still, all of it can be solved quite easily with proper sip proxies and
b2buas and efficient mediaproxies, see below.

On 08/30/2012 09:58 PM, rupert.or...@bt.com wrote:
 I have only ever deployed SER behind SBC. SBC is the most expensive 
 component. 

And there, the thinking often goes like it's the most expensive
component, so it must be the most secure one, and nobody got fired for
buying an Acme SBC so far. SBCs are marketed as firewalls for VoIP,
and security people love this idea.

 I am not involved with security, but I would not be allowed by my 
 organisation to deploy something like SER, with mysql details of usernames 
 and password on an internet address. 

In our systems we've a stateless kamailio instance running as load
balancer as single ingress/egress point in front of the proxies, b2buas,
app servers etc, and it only has a very small dbtext db for the
dispatcher table. No credentials of any kind exposed, and you can put it
on a separate node if you want.

 The SBC hides my addressing and only opens temp pinholes for both sip and 
 rtp. The main USP of the SBCs is the power to route many rtp streams 
 simultaneously. This also allows guard timers to check rtp streams dying so 
 no overcharging takes place. I don't think the hardware on a conventional 
 server will scale to the number of rtp streams i want to deal with. 

Our mediaproxy tells the b2bua internally via XMLRPC to tear down a call
if all streams of a call time out. Media relaying is done directly in
the kernel, you can handle thousands of parallel calls on a conventional
machine with a proper NIC, and you can add as many new nodes as you
like/need.

I guess u might say, why route all the rtps through a single  point of 
failure... there are other mechanisms to avoid overcharging,and also  can hide 
SER behind NAT... I agree, but SBC suppliers have moved along the value 
chainheader manipulation for interworking, eNum lookup, codec 
manipulationsagain all things that can be done on SER it comes down to 
a philosophy and way we do things...The other main thing SBC does is prevent 
attacks. i.e. SIP signalling attacks result in instant filtering from that 
address and thereby protects the service. I wouldn't know how to protect my 
SER if it was exposed to the internet, ...maybe there is a way...I just don't 
know it.

We do NAT handling on the kamailio lb instance, we do header
manipulation and codec filtering on an open source sbc (sems), and we do
DoS/DDoS protection also in kamailio lb, quite similar as described
here: http://kb.asipto.com/kamailio:usage:k31-sip-scanning-attack

The reason for us to use SBCs is really just to provide B2BUA features
like doing session timer handling, header/codec filtering, clean
topology hiding, and media stuff like music-on-hold and transcoding
etc., which all has been mentioned in different threads.

We still have customers who put an Acme SBC in front of all this, and
well, if they're comfortable with this and feel better, it's perfectly
fine. It more sometimes turns out though that the SBC behaves
incorrectly in quite a few scenarios, and then time-to-fix is most much
faster in an open source system rather than an expensive SBC appliance.

Andreas



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[SR-Users] [OT] sip:provider CE v2.6 is out!

[Andreas Granig]

Hi all,

I'm excited to announce the release of the Sipwise sip:provider
Community Edition v2.6, a free and open source turn-key platform, which
uses Kamailio, Sems and Asterisk in its core to provide a full-blown and
feature-rich VoIP soft-switch.

http://www.sipwise.com/news/announcements/spce-v2_6-release/

New core features are *full presence/xcap/im support* together with the
awesome Jitsi client, allowing you to deploy your own Skype-like service
for encryted voice/video communication, buddy lists, instant messaging,
screen sharing and remote desktop control. We also support *media
features for error announcements*, like busy, blocked, unknown number etc.

Check our VM images to get started within a couple of minutes and learn
what can be built with the power of Kamailio as a stateless
load-balancer and a stateful proxy/registrar, and Sems as an SBC and
application server.

http://www.sipwise.com/products/spce/quickinstall/

Thanks a lot to all developers, contributors and members of this mailing
list to make this possible and for helping us out!

Andreas



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[SR-Users] Wrong index on expires column in pua module

[Andreas Granig]

Hi guys,

In the pua table, there is a unique key on the expires column, which
seems wrong to me, because you'll sooner or later end up with different
pua entries having the same expiry time, no?

At least we ended up with messages like this in the logs, I think it's
caused by pua_reginfo tests:

ERROR: db_mysql [km_dbase.c:122]: driver error on query: Duplicate entry
'1345474412' for key 'expires_idx'
ERROR: core [db_query.c:210]: error while submitting query
ERROR: pua [pua.c:1171]: while inserting in db table pua

Any comments on that?

Andreas



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Re: [SR-Users] Kamailio and ACME

[Andreas Granig]

On 08/15/2012 08:23 PM, Fatima Chahrour wrote:
 Any help or ideas regarding the below?

Both lcr or dispatcher module can do what you want, check the docs at
http://kamailio.org/docs/modules/stable/ .

You need to get an idea how this stuff works by studying the example
config shipped with kamailio first though. Kamailio is a powerful proxy
which can do pretty much everything you want, but you need to tell it
explicitely *what* to do.

Andreas



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Re: [SR-Users] Kamailio and ACME

[Andreas Granig]

On 08/14/2012 11:29 AM, Fatima Chahrour~Vanrise Support wrote:
 I wanted to ask if Kamailio is compatible with ACME Packet or experianced by
 anyone in the list?
 and if there is any document that you recommend about such topic espacially
 in what is related to routing?

There is nothing special for interoperating Kamailio with ACME, it's
treated like any other SIP device. As always, it depends on what you
want to do.

Andreas



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Re: [SR-Users] Possible bad lcr module, after upgrade to 3.3.0 from 3.2

[Andreas Granig]

Hi,

Reviving this old thread, as we ran into the same issue as well. Turned
out it was a small glitch in passing a pointer to the size variable to
pcre, which was int instead of size_t, potentially causing stack
overflows on 64bit machines and messing with the rule_id, so the entry
got deactivated.

Richard has commited a fix today in trunk and 3.3, please try again to
see if it works now. If not, setting debug=5 and grepping for lcr in
the logs helps figuring out why the entry gets disabled.

Andreas

On 07/02/2012 08:48 AM, Juha Heinanen wrote:
 Alexey Mechanoshin writes:
 
 and if restart kamailio, again skipping from syslog:
 skipping disabled gw/rule = 1/5
 skipping disabled gw/rule = 3/5
 
 i tried again with your lcr table records (enclosed) and i do not get those
 skipping messages.  i cannot do anything more unless i'm able to
 reproduce your problem.
 
 -- juha
 
 
 
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Re: [SR-Users] Removing display-name with uac_replace_from

[Andreas Granig]

Hi,

The attached patch does the trick for me, but I'd rather like to have a
second opinion on it for any potential side effects before pushing it.

The point here is that the fixup function didn't change (beside the
name) between 3.1 and 3.3, however 3.1 used dedicated w_replace_from1()
and w_replace_from2(), whereas 3.3 uses one unified w_replace_from(),
which breaks the behavior the fixup function is providing.

Thanks,
Andreas

On 07/05/2012 12:32 AM, Andreas Granig wrote:
 Hi,
 
 On 07/04/2012 09:42 PM, Daniel-Constantin Mierla wrote:
 quick look into the sources shows that fixup makes the display parameter
 null if its length is 0 -- perhaps introduced in the patch that unified
 the fixup to work also for replacing the To header display. If you have
 time and urgent need, you can push a fix for it asap, otherwise I will
 look at it once getting some spare time out of traveling.
 
 Ok, I'll have a look at it tomorrow and push a fix.
 
 Andreas
 
 
 
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diff --git a/modules_k/uac/uac.c b/modules_k/uac/uac.c
index 8de09b1..763dd5a 100644
--- a/modules_k/uac/uac.c
+++ b/modules_k/uac/uac.c
@@ -435,14 +435,11 @@ static int fixup_replace_disp_uri(void** param, int param_no)
 			s.len += 2;
 		}
 	}
-	if(s.len!=0)
+	if(pv_parse_format(s ,model)0)
 	{
-		if(pv_parse_format(s ,model)0)
-		{
-			LM_ERR(wrong format [%s] for param no %d!\n, s.s, param_no);
-			pkg_free(s.s);
-			return E_UNSPEC;
-		}
+		LM_ERR(wrong format [%s] for param no %d!\n, s.s, param_no);
+		pkg_free(s.s);
+		return E_UNSPEC;
 	}
 	*param = (void*)model;
 


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[SR-Users] Removing display-name with uac_replace_from

[Andreas Granig]

Hi,

in kamailio 3.3, doing

  uac_replace_from(, $var(myuri));

doesn't remove the old Display part anymore (although the documentation
still says it should), the debug log is:

DEBUG: uac [uac.c:499]: dsp=(nil) (len=0) , uri=0x7fff87f6b840 (len=28)
DEBUG: uac [replace.c:324]: uri to replace [sip:a...@example.org]
DEBUG: uac [replace.c:325]: replacement uri is [sip:b...@example.org]
DEBUG: uac [replace.c:383]: encode
is=UEtlSyomCD8kIi8/SnlZMzNyS29QS2VLanBKeg-- len=40

So no hint that the old display is being removed. When doing

  uac_replace_from(foo, $var(myuri));

then there's an additional line like this:

DEBUG: uac [replace.c:284]: removing display [olddisplay]

which is missing when I put an empty string in the first parameter.

Andreas



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Re: [SR-Users] Removing display-name with uac_replace_from

[Andreas Granig]

Hi,

On 07/04/2012 09:42 PM, Daniel-Constantin Mierla wrote:
 quick look into the sources shows that fixup makes the display parameter
 null if its length is 0 -- perhaps introduced in the patch that unified
 the fixup to work also for replacing the To header display. If you have
 time and urgent need, you can push a fix for it asap, otherwise I will
 look at it once getting some spare time out of traveling.

Ok, I'll have a look at it tomorrow and push a fix.

Andreas



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Re: [SR-Users] How to access reason phrase of winning reply in failure_route

[Andreas Granig]

Hi,

On 07/02/2012 10:29 AM, Barthel Marco (CI/AFU1) wrote:
  how do I access the reason phrase of winning reply in failure_route?
 The reply code can be accessed via $T_reply_code. Is there something similar 
 for the reason phrase?

The tmx module exports $T_rpl(), which you can use like this to access
the reason: $T_rpl($rr)

Andreas



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[SR-Users] Binding arbitrary TLS socket

[Andreas Granig]

Hi,

Got some problems when I tried this, not sure how this is intended to work:

listen=tls:192.168.51.133:4343

When starting kamailio 3.3, I get the following error:

ERROR: core [tcp_main.c:2915]: ERROR: tcp_init: bind(10,
0x7f5eea45db24, 16) on 192.168.51.133:4343 : Address already in use
ERROR: tls [tls_init.c:314]: Error while initializing TCP part of TLS
socket 192.168.51.133:4343

What works though is if I explicitely bind a TCP socket one port below
the TLS socket, like this:

listen=tcp:192.168.51.133:4342
listen=tls:192.168.51.133:4343

Is this how it's intended to work? Is there a specific reason for this
behavior?

Andreas



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Re: [SR-Users] Binding arbitrary TLS socket

[Andreas Granig]

On 06/28/2012 04:35 PM, Andreas Granig wrote:
 Got some problems when I tried this, not sure how this is intended to work:
 
 listen=tls:192.168.51.133:4343
 
 When starting kamailio 3.3, I get the following error:
 
 ERROR: core [tcp_main.c:2915]: ERROR: tcp_init: bind(10,
 0x7f5eea45db24, 16) on 192.168.51.133:4343 : Address already in use
 ERROR: tls [tls_init.c:314]: Error while initializing TCP part of TLS
 socket 192.168.51.133:4343
 
 What works though is if I explicitely bind a TCP socket one port below
 the TLS socket, like this:
 
 listen=tcp:192.168.51.133:4342
 listen=tls:192.168.51.133:4343
 
 Is this how it's intended to work? Is there a specific reason for this
 behavior?

H, tried playing around with odd and even ports for tls, and now it
also works with the original approach from above (binding to TLS without
binding a TCP socket one port below that).

Maybe there was something wrong with my system somewhere (an old socket
still in use or something, although netstat didn't show up anything).
Does kamailio reuse old sockets for TCP/TLS?

Andreas



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Re: [SR-Users] About xcap xuid in kamailio

[Andreas Granig]

Hi,

On 06/13/2012 10:52 AM, Juha Heinanen wrote:
 my feeling is that there is no real commitment in jitsi for sip
 presence/xcap.  i found a presence bug long time ago and reported it on
 jitsi tracker.  so far nothing has happened.

For me, reporting a presence bug on the jitsi-users mailing list, along
with pointing out exactly where in the code and how to fix it eventually
worked. Did you get in touch with them on one of the lists?

In my opinion, it's quite important to work with the client guys,
because for some reason there are nearly no open source implementations
out there which properly support presence, so Jitsi is already one of
the better ones (what's the alternative?).

Andreas



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Re: [SR-Users] About xcap xuid in kamailio

[Andreas Granig]

Hi Juha,

On 06/13/2012 11:50 AM, Juha Heinanen wrote:
 For me, reporting a presence bug on the jitsi-users mailing list, along
 with pointing out exactly where in the code and how to fix it eventually
 worked. Did you get in touch with them on one of the lists?
 
 i don't remember if i send anything to mailing list.  the bug is this if
 you want to try out:
 
 http://java.net/jira/browse/JITSI-983

Looks like that has been addressed already. At least with svn trunk
verison, it's sending SIP-If-Match with Expires: 0. Could be worth
taking another try with a nightly 1.1 build?

Andreas



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Re: [SR-Users] About xcap xuid in kamailio

[Andreas Granig]

Hi Juha,

On 06/13/2012 05:07 PM, Juha Heinanen wrote:
 i did some more presence tests with jitsi. when i start jitsi, it
 registers its sip uri, subscribes to its own presence.winfo and presence
 of another uri, and publishes its own presence.
 
 when i then quit jitsi, it un-publishes its own presence, un-subscribes
 presence of the other uri, unregisters its own sip uri, and closes its
 tcp connection to kamailio.  it does unregistering/closing before it has
 received notify from kamailio presence server to the un-subscribe.  when
 presence server then tries to send the notify, it fails with syslog
 error, because tcp connection does not exist anymore.

Please send this to their us...@jitsi.java.net mailing list with a link
to this thread, so they realize that their implementation gets some
attraction.

Andreas



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Re: [SR-Users] Looking for RTP Proxy in TCP

[Andreas Granig]

Hi,

On 06/08/2012 03:23 PM, Aft nix wrote:
 http://www.mbdsys.com/foss/htproxy/file/f16c43f3c3c3/README

 it is kind of http proxy that can be used to tunnel udp packets. You need to
 have a client application supporting it, on the sip server side you don't
 need anything.
 
 Does this tunnel over HTTP? I mean the actual payload goes as
 payload of a http packet which goes over TCP?
 
 If i'm not wrong, for sending 30 bytes of actual voice data, you have
 send like 1K data?

Another approach we're using for example to bypass restrictive hotel
Wifis is having a VPN connected via TCP port 443, where you can tunnel
any kind of data then. I haven't seen anyone blocking this kind of
traffic, but who knows...

Andreas



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Re: [SR-Users] record calls audio and video

[Andreas Granig]

Hi,

On 06/05/2012 08:21 PM, Saul Waizer wrote:
 I am running Kamailio and rtpprox on ec2
 
 I do have a DBUG log for it, one interesting thing to point out: On the
 above line notice how I have 127.0.0.1, I changed this from my external
 ip address because i was getting the following on the rtpproxy log:
 
 Jun  5 15:42:55 ip-10-x-x-x rtpproxy[6526]: INFO:handle_command: new
 session NmJlNDNhOTA0YmYwZTZhYTEyMTQ4ZDIxY2IxNzQzYmM., tag 2c249068;1
 requested, type strong
 Jun  5 15:42:55 ip-10-x-x-x rtpproxy[6526]: ERR:create_twinlistener:
 can't bind to the IPv4 port 59794: Cannot assign requested address
 Jun  5 15:42:55 ip-10-x-x-x rtpproxy[6526]: ERR:handle_command: can't
 create listener
 Jun  5 15:42:55 ip-10-x-x-x rtpproxy[6526]: DBUG:doreply: sending reply
 6545_8 E10#012
 Jun  5 15:43:01 ip-10-x-x-x rtpproxy[6526]: DBUG:handle_command:
 received command 6556_12 Lc0,8,101
 NmJlNDNhOTA0YmYwZTZhYTEyMTQ4ZDIxY2IxNzQzYmM. some.ip 56536 2c249068;1
 c51afb04;1
 Jun  5 15:43:01 ip-10-x-x-x rtpproxy[6526]: INFO:handle_command: lookup
 request failed: session NmJlNDNhOTA0YmYwZTZhYTEyMTQ4ZDIxY2IxNzQzYmM.,
 tags 2c249068;1/c51afb04;1 not found
 
 This is expected because rtpproxy cant assign a port to the ec2 external
 ip instance, however I see that rtpproxy IS in fact trying to record but
 I just end up with a bunch of empty .rtp files.

Well, when starting rtpproxy, you need to tell it the address where it's
listening for rtp packets to relay (and record), and rtpproxy uses this
address as a response to kamailio, which in turn puts this information
into the SDP of the SIP messages and forwards it to the end devices.

If you put 127.0.0.1 there, then your end devices will end up sending
their RTP traffic to 127.0.0.1. Also, if you put the private IP of EC2
there (just to make rtpproxy happy and allow it to bind to this
address), it won't lead you to anywhere because that address is not
reachable from the end-devices. There is a patch floating around for
rtpproxy which allows you to set the advertised address (your public ip
mapped to your EC2 instance), which then will work with EC2.

In general, choosing EC2 for your SIP deployment is not a good idea if
you aren't 100% fit with kamailio, rtpproxy and SIP in general, because
EC2's NAT environment is quite tricky to handle in a SIP environment.

Andreas



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[SR-Users] Single-quoted vs double-quoted params in Path/Route

[Andreas Granig]

Hi guys,

When I implemented the Path module 6 years ago, I chose to enclose the
received-URI in double-quotes, like this:

Path: sip:1.2.3.4:5060;received=sip:1.2.3.5:1234;transport=tcp

This Path value is stored in usrloc by registrar module, and is
converted to a Route in a subsequent request. This worked fine so far,
because there's the logic in parser/parse_param.c to handle
double-quoted parameters.

Now while doing some interop tests with another platform, it turned out
that double-quotes are not allowed in URI params (see
https://lists.cs.columbia.edu/pipermail/sip-implementors/2008-May/019335.html,
and in the grammar the allowed chars in the unreserved definition are
alphanum and mark, where mark is only - / _ / . / ! / ~
/ * / ' / ( / ) ).

Anyways, to fully comply with RFC3261, I'd like to push a bugfix to path
module to use single-quotes instead, however it also needs a change in
the parser to use parse_quoted_param() also in case of single-quoted
params (which hasn't been handled at all until now).

My question now is if there are any concerns on your side with handling
single-quoted URI params the way we handle double-quoted ones? If we
were strict, we'd actually need to remove the handling for double-quoted
ones, but for backwards-compatibility I'd rather keep it there.

Comments?

Andreas



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Re: [SR-Users] Kamailio and Netann

[Andreas Granig]

Hi Patrice,

There are couple of characters in the URI which need to be escaped
properly. A correct R-URI would look like this:

sip:a...@cl1.oms.ser.v0.pftest.net;play=http%3A%2F%2F1xx.1xx.1xx.1xx%3A8080%2Ferrfile%2Ferr.wav;repeat=forever

There are string transformations {s.escape.param} and
{s.unescape.param} you can use (in case you get the full URI into your
config script, not sure whether the parser bails out first).

Andreas

On 06/04/2012 06:35 PM, patrice.bode...@orange.com wrote:
 Hello,
 
  
 
 I’am working on Kamailio version 3.2.2.
 
 I’m doing some test in interaction with Application server and media
 server.
 
 The applicationServer sends a INVITE sip to Media Server via the Kamailio.
 
 The R-URI has the format
 
 sip:a...@cl1.oms.ser.v0.pftest.net;play=http://1xx.1xx.1xx.1xx:8080/errfile/err.wav;repeat=forever
 
 But the parsing done by Kamailio of $ru is trunked to
 
 sip:a...@cl1.oms.ser.v0.pftest.net;play=http
 
  
 
 1)  Could you confirm if Kamailio supports Netann protocol ?
 
 2)  If not, how to route the INVITE with the correct R-URI format ?
 
  
 
 Thanks in advance.
 
  
 
 PatriceB
 
  
 
  
 
 _
 
 Ce message et ses pieces jointes peuvent contenir des informations 
 confidentielles ou privilegiees et ne doivent donc
 pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu 
 ce message par erreur, veuillez le signaler
 a l'expediteur et le detruire ainsi que les pieces jointes. Les messages 
 electroniques etant susceptibles d'alteration,
 France Telecom - Orange decline toute responsabilite si ce message a ete 
 altere, deforme ou falsifie. Merci.
 
 This message and its attachments may contain confidential or privileged 
 information that may be protected by law;
 they should not be distributed, used or copied without authorisation.
 If you have received this email in error, please notify the sender and delete 
 this message and its attachments.
 As emails may be altered, France Telecom - Orange is not liable for messages 
 that have been modified, changed or falsified.
 Thank you.
 
 
 
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Re: [SR-Users] Single-quoted vs double-quoted params in Path/Route

[Andreas Granig]

On 06/04/2012 07:29 PM, Iñaki Baz Castillo wrote:
 My question now is if there are any concerns on your side with handling
 single-quoted URI params the way we handle double-quoted ones? If we
 were strict, we'd actually need to remove the handling for double-quoted
 ones, but for backwards-compatibility I'd rather keep it there.
 
 IMHO the more friendly solution would be parse_quoted_param() to allow
 both single and double quoted values.
 
 Be liberal with receiving and strict while sending :)

Ack, that's how it's now implemented. I'll push the fix tomorrow if
there are no other objections.

Andreas



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Re: [SR-Users] Rtpproxy issue with connection information

[Andreas Granig]

Hi,

On 05/09/2012 02:40 PM, Openser Kamailio wrote:
 *Owner/Connection Information (o)*: doubango 1983 678901 IN IP4
 *172.27.170.984* 172.27.170.98
 *Connection Information (c)*: IN IP4  *172.27.170.984* 172.27.170.98

Could it be possible that you're calling rtpproxy_offer() twice?

Andreas



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Re: [SR-Users] htable feature request

[Andreas Granig]

On 04/18/2012 09:54 AM, Daniel-Constantin Mierla wrote:
 the freezing is next week, so it is possible if one codes it before.

Pushed to master. Sorry for the merge afterwards, forgot to rebase
before push. Need to get more into git :)

Andreas



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[SR-Users] htable feature request

[Andreas Granig]

Hi,

I'd like to ask for a feature request for htable. Would it be possible
to add an MI function to delete specific entries from an htable, e.g.
with sercmd htable.remove table key before freezing 3.3?

That'd be handy for example if you hold blocked IPs in htable, but want
to unblock some of them manually. At the moment, I only see the method
of restarting kamailio to whipe the whole table (and it'd only work if
dbmode=0).

Thanks in advance,
Andreas



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[SR-Users] [OT] sip:provider CE v2.5 is out!

[Andreas Granig]

Hi all,

I'm excited to announce the release of the Sipwise sip:provider
Community Edition v2.5, a free and open source turn-key platform, which
uses Kamailio, Sems and Asterisk in its core to provide a full-blown and
feature-rich VoIP soft-switch.

http://www.sipwise.com/news/announcements/spce-v2_5-release/

New core features are IPv6 and video support, serial call hunting and
time based routing, beside lots of other fixes and enhancements.

Check our VM images to get started within a couple of minutes and learn
what can be built with the power of Kamailio as a stateless
load-balancer and a stateful proxy/registrar, and Sems as an SBC and
application server.

Thanks a lot to all developers, contributors and members of this mailing
list to make this possible and for helping us out!

Andreas



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Re: [SR-Users] SEMS IVR issue

[Andreas Granig]

Hey Uri,

There's a dedicated mailing list for SEMS users:
http://lists.iptel.org/mailman/listinfo/sems

Regarding the docs, there are comprehensive Readmes for the different
modules in the doc/ folder. For IVR, there doesn't seem to be a
shared-object, because it looks like a set of python plugins. Check
doc/Readme.ivr.txt for instructions (can't tell you more about it as
I've never used that particular one, but it should give you a hint on
how to proceed further).

Andreas

On 03/29/2012 02:56 PM, Uri Shacked wrote:
 Hi,
  
 If anyone here is familiar with SEMS
 I installed it today, and could not figure out how to load the IVR
 application... there is no ivr.so file
 any ideas? there is so little info about it on the web...
  
 BR,
 Uri
 
 
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Re: [SR-Users] uac with more then one target?

[Andreas Granig]

Hi Karsten,

On 03/28/2012 12:52 PM, Karsten Horsmann wrote:
 Now some ask if we can act as UAC to get calls from them, for example
 authenticate
 to sipgate and feed the landline calls to our ivr system via kamailio.
 
 AFAIK the module UAC provides only one pair of user/password credentials.

I've never used UAC for authentication, but from the docs you can use
AVPs to set your credentials:
http://kamailio.org/docs/modules/3.2.x/modules_k/uac.html#auth-username-avp-id


So you'd fill that AVPs properly before doing the auth.

 Is that right? Have i here a chance to do the job with kamailio or
 must i use an second
 voip system for that?

In our sip:provider CE platforms we use Kamailio purely as a proxy and
use SEMS for that kind of things instead. You can provide peer
authentication credentials for subscribers, and SEMS will register in
their behalf on a 3rd-party-system. Then when a call comes in, kamailio
will map these for example sipgate-users to the local subscribers and
will deliver the call to whatever the routing logic is. Works like charm.

Andreas



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Re: [SR-Users] early media: How to change to tag

[Andreas Granig]

Hi,

On 03/28/2012 06:37 PM, Iñaki Baz Castillo wrote:
 2012/3/28 Min Wang ser.ba...@gmail.com:
 In order to properly proxy the msg to GW1, Kamailio seems need to change the
 to tag from B to A.
 
 Totally wrong. Multiple (early-)dialogs are 100% valid according to
 RFC 3261. If you find some SIP device failing when it receives
 multiple 180/183/200 responses with different To-tag, then drop it

I recently learned that for example Siemens switches implement Request
Disposition: no-fork defined in http://www.ietf.org/rfc/rfc3841.txt,
and if for some reason you decide to fork nonetheless on your side,
you'd probably want to do something about the different to-tags
(although you're violating against that specific RFC then). No idea how
such a device would react to not getting a to-tag at all by stripping it
out as Klaus suggested in another response, but at least that Siemens
switch doesn't bail out on getting different to-tags in provisional replies.

Andreas



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Re: [SR-Users] branches usage

[Andreas Granig]

Hi,

On 03/27/2012 04:46 PM, Uri Shacked wrote:
 In my case i need to change the header and then send it. 
 
 Here the case works if i make the changes after i sent the invite and
 got the reply. As i know the t_on_branch will work after the invite is
 sent... am i wrong?
 
 There are many questions to ask about the branches... when do they work
 in a serial manner, when parallel... where can i find it documented best?

Actually it's quite simple. If a request comes in and you modify it in a
normal route (e.g. calling rtpproxy_offer(), uac_replace_from(), adding
additional headers and stuff like that) and you send it out using e.g.
t_relay(), then in a serial scenario when you re-enter via a
failure-route, these changes will still be there (which is not always
what you want).

On the other hand, if you register a branch route via t_on_branch() and
do your changes there instead of prior to calling t_relay(), then these
changes from branch route will be rolled back, and you can do your
changes easily again in the same or another branch route starting with
your original message. This is important for example in serial call
forward scenarios, where the first destination might require an
rtpproxy, but the second doesn't, or when the first destination requires
different custom headers compare to the second.

Basically the same applies to parallel scenarios. If you modify your
message before calling lookup() and you've multiple contacts registered
for your destination, then your changes will apply for all branches
which are created by that. However if you move your modifications to a
branch route, then you can work on your messages individually, depending
on the branch.

Hope this clarifies it a bit.

Andreas



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