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




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