[SR-Users] RTPProxy issue?

Maxim Sobolev sobomax at sippysoft.com
Mon Mar 9 19:09:02 CET 2015


Igor, yes, I'd say give 2.0 a try and see if the problem is still there.
There were tons of changes, particularly in the rtp_resize subsystem.

Thanks!


On Sun, Mar 8, 2015 at 7:31 AM, Igor Potjevlesch <igor.potjevlesch at gmail.com
> wrote:

> Hello Maxim,
>
>
>
> I'm running legacy 1.2 or 1.4, not sure.
>
> I see in the latest code that the function is still there. Do you suggest
> to upgrade or there's a patch to make?
>
>
>
> Regards,
>
>
>
> Igor.
>
>
>
> *De :* sr-users [mailto:sr-users-bounces at lists.sip-router.org] *De la
> part de* Maxim Sobolev
> *Envoyé :* samedi 7 mars 2015 09:14
>
> *À :* Kamailio (SER) - Users Mailing List
> *Objet :* Re: [SR-Users] RTPProxy issue?
>
>
>
> Ah, ok, I see now. I did not realize you guys are using resizer. Which
> version of the software are you actually using? I.e. is it latest rel_2_0 /
> master, or some legacy 1.x code? We've done quite some revamping down
> there, so that it needs to be checked against the very latest code to make
> sure. Let us know.
>
> Thanks!
>
> On Mar 6, 2015 12:31 AM, "Igor Potjevlesch" <igor.potjevlesch at gmail.com>
> wrote:
>
> Hi Maxim,
>
>
>
> Hard to do because it's in production.
>
> I have a serious finding since yesterday on how this happened.
>
>
>
> My understanding is that the function "ts_less" returns FALSE into
> "rtp_resizer.c" because the timestamp between the two packets is > (1 <<
> 31) [for example: 3740425320].
>
> That's result in a drop of any following packets as I can see it into a
> capture.
>
>
>
> Regards,
>
>
>
> Igor.
>
>
>
> *De :* sr-users [mailto:sr-users-bounces at lists.sip-router.org] *De la
> part de* Maxim Sobolev
> *Envoyé :* vendredi 6 mars 2015 07:44
> *À :* Kamailio (SER) - Users Mailing List
> *Objet :* Re: [SR-Users] RTPProxy issue?
>
>
>
> Hi Igor, that's bit strange, since the rtpproxy is not checking any of the
> rtp flags including marker bit. It would help if you can post a tcpdump
> capture of the streams in question along with the log output of the
> rtpproxy running at the "dbug" level. Thanks!
>
> On Mar 5, 2015 5:54 AM, "Igor Potjevlesch" <igor.potjevlesch at gmail.com>
> wrote:
>
> I reviewed again a call trace and I can be more precise: a RTP packet
> comes with a new SSRC and the Marker bit set to "True". This packet is
> properly forwarded.
>
>
>
> Then, just after this packet, another RTP packet containing a new SSRC
> with the huge timestamp and the Marker bit set to "True" is coming from the
> UA.
>
> The RTPProxy stops forward since this packet.
>
>
>
> Regards,
>
>
>
> Igor.
>
>
>
> *De :* Igor Potjevlesch [mailto:igor.potjevlesch at gmail.com]
> *Envoyé :* jeudi 5 mars 2015 11:34
> *À :* miconda at gmail.com; 'Kamailio (SER) - Users Mailing List'
> *Objet :* RE: [SR-Users] RTPProxy issue?
>
>
>
> Hello,
>
>
>
> Thank you.
>
>
>
> Just to let you know, the RTPProxy is running in bridging mode.
>
> Regards,
>
>
>
> Igor.
>
>
>
> *De :* sr-users [mailto:sr-users-bounces at lists.sip-router.org
> <sr-users-bounces at lists.sip-router.org>] *De la part de*
> Daniel-Constantin Mierla
> *Envoyé :* jeudi 5 mars 2015 09:33
> *À :* Kamailio (SER) - Users Mailing List
> *Objet :* Re: [SR-Users] RTPProxy issue?
>
>
>
> Hello,
>
> maybe Maxim (cc-ed) will be able to provide more insights.
>
> Cheers,
> DAniel
>
> On 04/03/15 16:59, Igor Potjevlesch wrote:
>
> Hello,
>
>
>
> I discovered an issue related to the handling of "timestamp" and/or
> "Marker bit" with rtpproxy (I use the latest Extension 20081224).
>
>
>
> The call-flow is the following: one UA places a call to A and put this
> call on hold. Then, the same UA call another number B. Individual streams
> are ok.
>
> When the UA tries to transfer A with B, the RTPProxy receive a RTP packet
> with a huge timestamp and the Marker bit set to "True".
>
>
>
> Just after this RTP packet, RTPProxy stop forward the RTP packets from A
> to B. B to C is still working.
>
>
>
> Anyone have an idea?
>
> Regards,
>
>
>
> Igor.
>
>
>
> _______________________________________________
>
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>
> sr-users at lists.sip-router.org
>
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
> --
>
> Daniel-Constantin Mierla
>
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>
> Kamailio World Conference, May 27-29, 2015
>
> Berlin, Germany - http://www.kamailioworld.com
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>


-- 
Maksym Sobolyev
Sippy Software, Inc.
Internet Telephony (VoIP) Experts
Tel (Canada): +1-778-783-0474
Tel (Toll-Free): +1-855-747-7779
Fax: +1-866-857-6942
Web: http://www.sippysoft.com
MSN: sales at sippysoft.com
Skype: SippySoft
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20150309/fea6c6e7/attachment.html>


More information about the sr-users mailing list