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@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@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@freeswitch.comhttps://freeswitch.com
Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.c...
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswit...
_________________________________________________________________________
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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
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