Error 1004 reason=Incompatible SDP

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Sergio Charpinel

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Apr 13, 2018, 2:50:42 PM4/13/18
to BigBlueButton-dev
I've setup a local bigbluebutton 1.1 server with HTTPs but I am getting the following 1004 error when I try to connect the mic:

Detected the following WebRTC issue: Error 1004: Failure on call (reason=Incompatible SDP). Do you want to try Flash instead?

I also tried to use https://demo.bigbluebutton.org/b and got the same error. So I'm suspecting something with my browsers or Windows configuration. Tested with Chrome and Firefox using a laptop and a workstation.

Here the relevant part of my browser console log using my server. I got the almost the same message with demo.bigbluebutton.com.

Via: SIP/2.0/WSS me4i8q2epf9f.invalid;branch=z9hG4bK5753952
Max-Forwards: 70
From: "4heh57bnh4ub_1-bbbID-teste" <sip:4heh57bnh4ub_...@conferencia.example.com>;tag=ae2qt8bf8k
Call-ID: bk8cah2ds2si35j4g182
CSeq: 5041 INVITE
Contact: <sip:g549...@me4i8q2epf9f.invalid;transport=ws;ob>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Content-Type: application/sdp
Supported: outbound
User-Agent: BigBlueButton
Content-Length: 1677

v=0
o=- 493636722730294963 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS I6eQ5zPd2ym8PyVrBkFty1SpozwZoyDB99CZ
m=audio 51937 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 10.205.64.250
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:2656445765 1 udp 2122260223 10.205.64.250 51937 typ host generation 0 network-id 2
a=candidate:2892323760 1 udp 2122194687 10.205.140.129 51938 typ host generation 0 network-id 1 network-cost 10
a=candidate:3503675829 1 tcp 1518280447 10.205.64.250 9 typ host tcptype active generation 0 network-id 2
a=candidate:3806766912 1 tcp 1518214911 10.205.140.129 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=ice-ufrag:0MEf
a=ice-pwd:rrzqKGR1HDs6dvSWcBOGssZ7
a=fingerprint:sha-256 39:93:B2:DF:57:B9:5F:E8:2D:15:B8:93:AD:CC:53:9D:64:46:B2:8C:41:A2:13:37:A4:F3:E5:0E:25:92:A8:29
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:2563993322 cname:1Si7quEZ686hLvF4
a=ssrc:2563993322 msid:I6eQ5zPd2ym8PyVrBkFty1SpozwZoyDB99CZ 005fa32f-8cfc-49d7-b37e-4aab7f4a1c06
a=ssrc:2563993322 mslabel:I6eQ5zPd2ym8PyVrBkFty1SpozwZoyDB99CZ
a=ssrc:2563993322 label:005fa32f-8cfc-49d7-b37e-4aab7f4a1c06


sip.js?v=601:2884 Fri Apr 13 2018 13:33:06 GMT-0300 (E. South America Standard Time) | sip.transport | received WebSocket text message:

SIP/2.0 100 Trying
Via: SIP/2.0/WSS me4i8q2epf9f.invalid;branch=z9hG4bK5753952;received=10.205.4.12;rport=59026
From: "4heh57bnh4ub_1-bbbID-teste" <sip:4heh57bnh4ub_...@conferencia.example.com>;tag=ae2qt8bf8k
Call-ID: bk8cah2ds2si35j4g182
CSeq: 5041 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.6.15+git~20170214T173745Z~bec45386a2~64bit
Content-Length: 0



bbb_webrtc_bridge_sip.js?v=601:427 call progress: SIP/2.0 100 Trying
Via: SIP/2.0/WSS me4i8q2epf9f.invalid;branch=z9hG4bK5753952;received=10.205.4.12;rport=59026
From: "4heh57bnh4ub_1-bbbID-teste" <sip:4heh57bnh4ub_...@conferencia.example.com>;tag=ae2qt8bf8k
Call-ID: bk8cah2ds2si35j4g182
CSeq: 5041 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.6.15+git~20170214T173745Z~bec45386a2~64bit
Content-Length: 0


sip.js?v=601:2884 Fri Apr 13 2018 13:33:06 GMT-0300 (E. South America Standard Time) | sip.transport | received WebSocket text message:

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/WSS me4i8q2epf9f.invalid;branch=z9hG4bK5753952;received=10.205.4.12;rport=59026
Max-Forwards: 69
From: "4heh57bnh4ub_1-bbbID-teste" <sip:4heh57bnh4ub_...@conferencia.example.com>;tag=ae2qt8bf8k
To: <sip:9196...@conferencia.example.com>;tag=2cQgFUapvejXj
Call-ID: bk8cah2ds2si35j4g182
CSeq: 5041 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.6.15+git~20170214T173745Z~bec45386a2~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0
Remote-Party-ID: "919676962" <sip:9196...@conferencia.example.com>;party=calling;privacy=off;screen=no



sip.js?v=601:2884 Fri Apr 13 2018 13:33:06 GMT-0300 (E. South America Standard Time) | sip.transport | sending WebSocket message:

Via: SIP/2.0/WSS me4i8q2epf9f.invalid;branch=z9hG4bK5753952
To: <sip:9196...@conferencia.example.com>;tag=2cQgFUapvejXj
From: "4heh57bnh4ub_1-bbbID-teste" <sip:4heh57bnh4ub_...@conferencia.example.com>;tag=ae2qt8bf8k
Call-ID: bk8cah2ds2si35j4g182
Content-Length: 0
CSeq: 5041 ACK



bbb_webrtc_bridge_sip.js?v=601:431 call failed with cause: Incompatible SDP
bbb_webrtc_bridge_sip.js?v=601:116 Stopping webrtc audio test



Thanks in advance.

Chad Pilkey

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Apr 13, 2018, 4:35:15 PM4/13/18
to BigBlueButton-dev
I think it's failing because all of your candidates are local IP addresses and FreeSWITCH automatically rejects the call when that is the case. It looks like you maybe aren't able to contact the STUN server to get your public IP. Could be a network issue or a computer issue. We use "stun.freeswitch.org" by default.


On Friday, April 13, 2018 at 2:50:42 PM UTC-4, Sergio Charpinel wrote:
I've setup a local bigbluebutton 1.1 server with HTTPs but I am getting the following 1004 error when I try to connect the mic:

Detected the following WebRTC issue: Error 1004: Failure on call (reason=Incompatible SDP). Do you want to try Flash instead?

I also tried to use https://demo.bigbluebutton.org/b and got the same error. So I'm suspecting something with my browsers or Windows configuration. Tested with Chrome and Firefox using a laptop and a workstation.

Here the relevant part of my browser console log using my server. I got the almost the same message with demo.bigbluebutton.com.

Via: SIP/2.0/WSS me4i8q2epf9f.invalid;branch=z9hG4bK5753952
Max-Forwards: 70
Call-ID: bk8cah2ds2si35j4g182
CSeq: 5041 INVITE
Contact: <sip:g549p878@me4i8q2epf9f.invalid;transport=ws;ob>
Call-ID: bk8cah2ds2si35j4g182
CSeq: 5041 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.6.15+git~20170214T173745Z~bec45386a2~64bit
Content-Length: 0



bbb_webrtc_bridge_sip.js?v=601:427 call progress: SIP/2.0 100 Trying
Via: SIP/2.0/WSS me4i8q2epf9f.invalid;branch=z9hG4bK5753952;received=10.205.4.12;rport=59026
Call-ID: bk8cah2ds2si35j4g182
CSeq: 5041 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.6.15+git~20170214T173745Z~bec45386a2~64bit
Content-Length: 0


sip.js?v=601:2884 Fri Apr 13 2018 13:33:06 GMT-0300 (E. South America Standard Time) | sip.transport | received WebSocket text message:

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/WSS me4i8q2epf9f.invalid;branch=z9hG4bK5753952;received=10.205.4.12;rport=59026
Max-Forwards: 69
From: "4heh57bnh4ub_1-bbbID-teste" <sip:4heh57bnh4ub_1-bbbID-te...@conferencia.example.com>;tag=ae2qt8bf8k
Call-ID: bk8cah2ds2si35j4g182
CSeq: 5041 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.6.15+git~20170214T173745Z~bec45386a2~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0
Remote-Party-ID: "919676962" <sip:919676962@conferencia.example.com>;party=calling;privacy=off;screen=no



sip.js?v=601:2884 Fri Apr 13 2018 13:33:06 GMT-0300 (E. South America Standard Time) | sip.transport | sending WebSocket message:

Via: SIP/2.0/WSS me4i8q2epf9f.invalid;branch=z9hG4bK5753952
From: "4heh57bnh4ub_1-bbbID-teste" <sip:4heh57bnh4ub_1-bbbID-te...@conferencia.example.com>;tag=ae2qt8bf8k
Call-ID: bk8cah2ds2si35j4g182
Content-Length: 0
CSeq: 5041 ACK



bbb_webrtc_bridge_sip.js?v=601:431 call failed with cause: Incompatible SDP
bbb_webrtc_bridge_sip.js?v=601:116 Stopping webrtc audio test



Thanks in advance.
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