[SR-Users] rtcp-mux is missing in the OK for WebRTC-Calls

Fri Jul 10 10:59:59 CEST 2020

Hey guys,

I'm trying to configure a Kamailio to work with a browser softphone based on SIPJS using WebRTC.
So far it works great on Firefox but have a specific problem with chrome, when I want to make call from the softphone to another extension.
After anwsering the call Chrome/the softphone sends a BYE immediately, because this line "a=rtcp-mux" is missing in the OK.

The Kamailio is a proxy. Behind the Kamailio there is an Asterisk, which is responsible for the pbx-features.

Those are my rtpengine Flags for the Invite:

rtpengine_manage: replace-origin replace-session-connection trust-address via-branch=extra rtcp-mux-demux DTLS=off SDES-on ICE=remove RTP/AVP

And those are the flags for the response, in this case the OK:

rtpengine_manage: replace-origin replace-session-connection rtcp-mux-offer rtcp-mux-accept generate-mid DTLS=off SDES-on ICE=force RTP/SAVPF direction=internal direction=external loop-protect

It seems that the Kamailio ignores ths "rtcp-mux-offer rtcp-mux-accept" in the response. Can you help me get it to work?

Here is the SIP-Dialog. The call is from extension 201 to the extension 2.

INVITE:

INVITE sip:2 at mydomain SIP/2.0
Via: SIP/2.0/WSS g4n0lpfirgpv.invalid;branch=z9hG4bK5104892
To: <sip:2 at mydomain.com>
From: <sip:201 at mydomain.com>;tag=aqplr71k05
CSeq: 2 INVITE
Call-ID: sde09v49f8gqtt59oddn
Max-Forwards: 70
Proxy-Authorization: Digest algorithm=MD5, username="201", realm="mydomain.com", nonce="sfdsdfdsfsdfdsfsdfaseqww", uri="sip:2 at mydomain.com", response="81rewtrega23423r"
Contact: <sip:kf6iduon at g4n0lpfirgpv.invalid;transport=ws;ob>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: SIPJS
Content-Type: application/sdp
Content-Length: 1916

v=0
o=- 5826889093965459811 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=msid-semantic: WMS Jk5GQwzPaVSTxyIZER7RBqkkMNWCdmKmMdyO
m=audio 54274 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 mycomputerIP
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:1577908739 1 udp 2113937151 efd6d297-d186-4c07-87d8-933e5846a82d.local 54274 typ host generation 0 network-cost 999
a=candidate:842163049 1 udp 1677729535 mycomputerIP 54274 typ srflx raddr 0.0.0.0 rport 0 generation 0 network-cost 999
a=ice-ufrag:Ktxl
a=ice-pwd:x5xDUb41GeOXAcHNQlra4yUN
a=ice-options:trickle
a=fingerprint:sha-256 2C:7C:32:AF:34:A3:9D:AE:C7:FD:92:68:DD:D8:AB:82:DB:F0:32:51:14:97:20:60:66:5C:2F:CF:B7:98:B8:A8
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:Jk5GQwzPaVSTxyIZER7RBqkkMNWCdmKmMdyO dc187be3-4cd0-4129-98e4-f804cd8d2c94
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:2376769177 cname:RgPkYyxDrEJTpmO2
a=ssrc:2376769177 msid:Jk5GQwzPaVSTxyIZER7RBqkkMNWCdmKmMdyO dc187be3-4cd0-4129-98e4-f804cd8d2c94
a=ssrc:2376769177 mslabel:Jk5GQwzPaVSTxyIZER7RBqkkMNWCdmKmMdyO
a=ssrc:2376769177 label:dc187be3-4cd0-4129-98e4-f804cd8d2c94

RINGING:

SIP/2.0 180 Ringing
Via: SIP/2.0/WSS g4n0lpfirgpv.invalid;rport=60196;received=mycomputerIP;branch=z9hG4bK5104892
Record-Route: <sip:internalIP;r2=on;lr=on;ftag=aqplr71k05;nat=yes;rtp=ws>
Record-Route: <sip:externalIP;transport=ws;r2=on;lr=on;ftag=aqplr71k05;nat=yes;rtp=ws>
From: <sip:201 at mydomain.com>;tag=aqplr71k05
To: <sip:2 at mydomain.com>;tag=as64fe0fc8
Call-ID: sde09v49f8gqtt59oddn
CSeq: 2 INVITE
Server: myserver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <mycontact>
P-Asserted-Identity: "Phone 2" <sip:2 at mydomain.com>
Content-Length: 0

OK:

SIP/2.0 200 OK
Via: SIP/2.0/WSS g4n0lpfirgpv.invalid;rport=60196;received=mycomputerIP;branch=z9hG4bK5104892
Record-Route: <sip:internal-ip;r2=on;lr=on;ftag=aqplr71k05;nat=yes;rtp=ws>
Record-Route: <sip:external-ip;transport=ws;r2=on;lr=on;ftag=aqplr71k05;nat=yes;rtp=ws>
From: <sip:201 at mydomain.com>;tag=aqplr71k05
To: <sip:2 at mydomain.com>;tag=as64fe0fc8
Call-ID: sde09v49f8gqtt59oddn
CSeq: 2 INVITE
Server: myserver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <mycontact>
P-Asserted-Identity: "Phone 2" <sip:2 at mydomain.com>
Content-Type: application/sdp
Content-Length: 806

v=0
o=root 882840402 882840402 IN IP4 externalIP
s=Asterisk PBX 11.22.0
c=IN IP4 externalIP
t=0 0
a=rtpengine:4d633e3022f5
m=audio 16416 RTP/SAVPF 111 9 8 0 126
a=maxptime:60
a=mid:0
a=rtpmap:111 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:111 maxplaybackrate=16000; stereo=0; sprop-stereo=0; useinbandfec=0
a=fmtp:126 0-16
a=sendrecv
a=rtcp:16417
a=setup:active
a=fingerprint:sha-1 B1:67:4B:B8:47:89:E8:49:CD:DD:F8:FF:41:5C:83:72:D9:DE:4D:45
a=ptime:20
a=ice-ufrag:WBZN8m7c
a=ice-pwd:mKVR6Jv0G0pvrTv7OfEm2wPW9Q
a=ice-options:trickle
a=candidate:3kddB3zBUV2n2jt0 1 UDP 2130706431 externalIP 16416 typ host
a=candidate:3kddB3zBUV2n2jt0 2 UDP 2130706430 externalIP 16417 typ host
a=end-of-candidates

I hope, you can help me with this. If you have further question, I will try to answer them as best I can.

Greetings,

Ben