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
What is the call flow and devices involved? You shouldn't be getting the webrtc SDP if your bridge line and settings are correct, can you elaborate? Also that second part looks like a bug those should go on JIRA. On Thu, Sep 24, 2015 at 9:38 AM, Serge Yuriev <me@nevian.org> wrote:
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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participants (2)
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Brian West
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Serge Yuriev