[Freeswitch-users] Calls dropping due to SDP change

Mon Dec 28 18:19:26 UTC 2020

I've seen this problem before: the remote end is preferring PCMA over PCMU
when it does the session timer, If you're talking to any providers in the
EU, disallow PMCU always.

Can you provide a full sip trace of this?

On Mon, Dec 28, 2020 at 9:16 AM Nicola von Thadden <nico at vthadden.de> wrote:

> Hi,
>
> yes, we are using HD audio there. Deutsche Telekom is codec transparent,
> they even pass opus and other codecs since a while.
> So disabling any codecs but G.711 is not really a solution, especially
> since DT is the biggest provider in Germany and that issue might not only
> affect my setup but in theory all other Freeswitch in Germany here which
> might want to call a DT landline or mobile number.
>
> Is there a way to make freeswitch stop mixing up the SDP?
>
> Nico
>
> On 12/28/20 12:04 AM, Brian : wrote:
>
> Unless you're doing hd audio on these calls your solution is to just use
> g711 on calls into this provider.
>
> On Saturday, December 26, 2020, Nicola von Thadden <nico at vthadden.de>
> wrote:
> > Hi,
> >
> > I'm currently investigating reproducable call drops when calling
> > mobile-phone numbers from Deutsche Telekom (T-Mobile Germany).
> > My provider has a NGN (SIP) Interconnection with Deutsche Telekom and is
> > transparend for the messages involved here (although IP rewriting is
> > happening).
> >
> > The calls contain following SDP in the invite from my FreeSwitch towards
> > my provider:
> >
> v=0
> >
> > o=FreeSWITCH 1608967203 1608967204 IN IP4 *redacted*
> >
>
> >
> >
> s=FreeSWITCH
> >
> > c=IN IP4  *redacted*
> >
>
> >
> > t=0
> >
> 0
> >
> > m=audio 32256 RTP/AVP 9 8 0
> >
> 101
> >
> > a=rtpmap:9
> >
> G722/8000
> >
> > a=rtpmap:8
> >
> PCMA/8000
> >
> > a=rtpmap:0
> >
> PCMU/8000
> >
> > a=rtpmap:101
> >
> telephone-event/8000
> >
> > a=fmtp:101
> >
> 0-16
> >
> > a=ptime:20
> >
> > Once the call is established, FS sends a re-invite after 50% of the
> > expiration timer is elapsed, 15 minutes in this case. The re-invite
> > contains a slightly modified SDP:
> >
> v=0
> >
> > o=FreeSWITCH 1608967203 1608967204 IN IP4
> >
> *redacted*
> >
> >
> s=FreeSWITCH
> >
> > c=IN IP4
> >
> *redacted*
> >
> > t=0
> >
> 0
> >
> > m=audio 32256 RTP/AVP 8 101 9
> >
> 0
> >
> > a=rtpmap:8
> >
> PCMA/8000
> >
> > a=rtpmap:101
> >
> telephone-event/8000
> >
> > a=fmtp:101
> >
> 0-16
> >
> > a=rtpmap:9
> >
> G722/8000
> >
> > a=rtpmap:0
> >
> PCMU/8000
> >
> > a=ptime:20
> >
> > The new codec on position 1 (PCMA) is not necessary the chosen one for
> > the session, that call was using G.722 (verified via 'show channels').
> >
> > Telekom does not like my SDP change and responds with:
> > SIP/2.0 488 SDP Parameter Error In SIP
> >
> Request
> >
> >
> > The freeswitch console only logs:
> > 2020-12-26 16:32:49.578925 [DEBUG] sofia.c:7326 Channel
> > sofia/external/*redacted*entering state [calling][0]
> > 2020-12-26 16:32:49.618925 [DEBUG] sofia.c:7319 Channel
> > sofia/external/*redacted* skipping state [ready][488]
> >
> > The call is disconnected 15 minutes later because the session timer has
> > expired:
> > 2020-12-26 16:47:53.838925 [NOTICE] sofia.c:1089 Hangup
> > sofia/external/*redacted* [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
> >
> > This does not happen when I only allow PCMA (or any other codec) since
> > the SDP can't get mixed up.
> >
> > I have reproduced that behaviour with
> > 1.10.5~release~6~25569c1631~buster-1~buster+1 and
> > 1.6.20~37~987c9b9-1~jessie+1.
> >
> > Do you have any idea why FS changes the SDP (without reason?) and what I
> > can do about it?
> >
> > Thanks
> > Nico
> >
> > _________________________________________________________________________
> >
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>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time https://signalwire.community
>
> Professional FreeSWITCH Servicessales at freeswitch.comhttps://freeswitch.com
>
> Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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