[Freeswitch-users] Calls dropping due to SDP change

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