Unless you're doing hd audio on these calls your solution is to just use g711 on calls into this provider. <br><br>On Saturday, December 26, 2020, Nicola von Thadden <<a href="mailto:nico@vthadden.de">nico@vthadden.de</a>> wrote:<br>> Hi,<br>><br>> I'm currently investigating reproducable call drops when calling<br>> mobile-phone numbers from Deutsche Telekom (T-Mobile Germany).<br>> My provider has a NGN (SIP) Interconnection with Deutsche Telekom and is<br>> transparend for the messages involved here (although IP rewriting is<br>> happening).<br>><br>> The calls contain following SDP in the invite from my FreeSwitch towards<br>> my provider:<br>> v=0                                                                                                                                                                                                                                                                                                                         <br>><br>> o=FreeSWITCH 1608967203 1608967204 IN IP4 *redacted*<br>>                                                                                                                                                                                                                                                                     <br>><br>> s=FreeSWITCH                                                                                                                                                                                                                                                                                                                <br>><br>> c=IN IP4  *redacted*<br>>                                                                                                                                                                                                                                                                                                   <br>><br>> t=0<br>> 0                                                                                                                                                                                                                                                                                                                       <br>><br>> m=audio 32256 RTP/AVP 9 8 0<br>> 101                                                                                                                                                                                                                                                                                             <br>><br>> a=rtpmap:9<br>> G722/8000                                                                                                                                                                                                                                                                                                        <br>><br>> a=rtpmap:8<br>> PCMA/8000                                                                                                                                                                                                                                                                                                        <br>><br>> a=rtpmap:0<br>> PCMU/8000                                                                                                                                                                                                                                                                                                        <br>><br>> a=rtpmap:101<br>> telephone-event/8000                                                                                                                                                                                                                                                                                           <br>><br>> a=fmtp:101<br>> 0-16                                                                                                                                                                                                                                                                                                             <br>><br>> a=ptime:20  <br>><br>> Once the call is established, FS sends a re-invite after 50% of the<br>> expiration timer is elapsed, 15 minutes in this case. The re-invite<br>> contains a slightly modified SDP:<br>> v=0                                                                                                                                                                                                                                                                                                                         <br>><br>> o=FreeSWITCH 1608967203 1608967204 IN IP4<br>> *redacted*                                                                                                                                                                                                                                                                     <br>><br>> s=FreeSWITCH                                                                                                                                                                                                                                                                                                                <br>><br>> c=IN IP4<br>> *redacted*                                                                                                                                                                                                                                                                                                      <br>><br>> t=0<br>> 0                                                                                                                                                                                                                                                                                                                       <br>><br>> m=audio 32256 RTP/AVP 8 101 9<br>> 0                                                                                                                                                                                                                                                                                             <br>><br>> a=rtpmap:8<br>> PCMA/8000                                                                                                                                                                                                                                                                                                        <br>><br>> a=rtpmap:101<br>> telephone-event/8000                                                                                                                                                                                                                                                                                           <br>><br>> a=fmtp:101<br>> 0-16                                                                                                                                                                                                                                                                                                             <br>><br>> a=rtpmap:9<br>> G722/8000                                                                                                                                                                                                                                                                                                        <br>><br>> a=rtpmap:0<br>> PCMU/8000                                                                                                                                                                                                                                                                                                        <br>><br>> a=ptime:20   <br>><br>> The new codec on position 1 (PCMA) is not necessary the chosen one for<br>> the session, that call was using G.722 (verified via 'show channels').<br>><br>> Telekom does not like my SDP change and responds with:<br>> SIP/2.0 488 SDP Parameter Error In SIP<br>> Request                                                                                                                                                                                                                                                                              <br>><br>><br>> The freeswitch console only logs:<br>> 2020-12-26 16:32:49.578925 [DEBUG] sofia.c:7326 Channel<br>> sofia/external/*redacted*entering state [calling][0]<br>> 2020-12-26 16:32:49.618925 [DEBUG] sofia.c:7319 Channel<br>> sofia/external/*redacted* skipping state [ready][488]<br>><br>> The call is disconnected 15 minutes later because the session timer has<br>> expired:<br>> 2020-12-26 16:47:53.838925 [NOTICE] sofia.c:1089 Hangup<br>> sofia/external/*redacted* [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]<br>><br>> This does not happen when I only allow PCMA (or any other codec) since<br>> the SDP can't get mixed up.<br>><br>> I have reproduced that behaviour with<br>> 1.10.5~release~6~25569c1631~buster-1~buster+1 and<br>> 1.6.20~37~987c9b9-1~jessie+1.<br>><br>> Do you have any idea why FS changes the SDP (without reason?) and what I<br>> can do about it?<br>><br>> Thanks<br>> Nico<br>><br>> _________________________________________________________________________<br>><br>> The FreeSWITCH project is sponsored by SignalWire <a href="https://signalwire.com">https://signalwire.com</a><br>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.<br>> Build your next product on our scalable cloud platform.<br>><br>> Join our online community to chat in real time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