Hi, I’m trying to disable any mentions about WebRTC/DTLS in SDP. My PBX can’t understand them and drops calls. I use pre_answer app and get such SDP towards my PBX send 1698 bytes to udp/[10.0.64.63]:6060 at 20:02:39.595098: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.64.63:6060;branch=z9hG4bK300267909a88 From: "IT, ????? ??????" <sip:12550@10.0.64.63>;tag=357758~27154efa-6325-45a2-9e47-67e5d9302ebc-249340814 To: <sip:17000@10.0.64.100>;tag=73ccUeacZSQ7c Call-ID: bb282700-5f91a0af-24648-3f40000a@10.0.64.63 CSeq: 101 INVITE Contact: <sip:mod_sofia@10.0.64.100:6060> RSeq: 145774175 User-Agent: FreeSWITCH-mod_sofia/1.6.0+git~20150903T203652Z~6762f14140~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, PRACK, NOTIFY Require: 100rel Supported: precondition, 100rel, timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 925 v=0 o=FreeSWITCH 1442395963 1442395964 IN IP4 10.0.64.100 s=FreeSWITCH c=IN IP4 10.0.64.100 t=0 0 a=msid-semantic: WMS wp2SlBEpzCNtEIOknB2WO45EEsyGM2B8 m=audio 26996 RTP/SAVPF 18 9 8 0 3 101 13 a=rtpmap:18 G729/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=rtpmap:13 CN/8000 a=fingerprint:sha-256 09:0E:1F:D3:0D:FD:F9:54:AA:FF:AD:7F:CC:0A:85:94:D6:16:38:9F:94:43:3F:78:AC:E8:FC:8B:2B:0C:83:CA a=setup:actpass a=rtcp-mux a=rtcp:26996 IN IP4 10.0.64.100 a=ssrc:1443086239 cname:huRtVlIlOmEjUjt8 a=ssrc:1443086239 msid:wp2SlBEpzCNtEIOknB2WO45EEsyGM2B8 a0 a=ssrc:1443086239 mslabel:wp2SlBEpzCNtEIOknB2WO45EEsyGM2B8 a=ssrc:1443086239 label:wp2SlBEpzCNtEIOknB2WO45EEsyGM2B8a0 a=ice-ufrag:LmUqgPUL8iiKbJJj a=ice-pwd:3qCPCb0hnvq8FJ6gVQw2vhEu a=candidate:7296319604 1 udp 659136 10.0.64.100 26996 typ host generation 0 a=ptime:20 ———————————————————————————————————— If I use webrtc_enable_dtls=false, FS don’t send any codecs send 932 bytes to udp/[10.0.64.63]:6060 at 18:25:17.885114: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.64.63:6060;branch=z9hG4bK311ba4379c394 From: "IT, ????? ??????" <sip:12550@10.0.64.63>;tag=364113~27154efa-6325-45a2-9e47-67e5d9302ebc-249450749 To: <sip:17000@10.0.64.100>;tag=FcQj96g8709vS Call-ID: 4b777200-5fa1db5d-24eb6-3f40000a@10.0.64.63 CSeq: 101 INVITE Contact: <sip:mod_sofia@10.0.64.100:6060> RSeq: 485241663 User-Agent: FreeSWITCH-mod_sofia/1.6.0+git~20150903T203652Z~6762f14140~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, PRACK, NOTIFY Require: 100rel Supported: precondition, 100rel, timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 158 v=0 o=FreeSWITCH 1442476665 1442476666 IN IP4 10.0.64.100 s=FreeSWITCH c=IN IP4 10.0.64.100 t=0 0 a=msid-semantic: WMS TqexzJilLgNDKOovoQ0kCWDMDNGQpja8 ———————————————————————————————————— How I can get rid of ICE/WebRTC/DTLS parts only? -- Serge S. Yuriev