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@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 Services
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https://freeswitch.com

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