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sip.SessionDescriptionHandler | SessionDescriptionHandler.setDescription failed - InvalidAccessError: Failed to execute 'setRemoteDescription' on 'RTCPeerConnection': Failed to set remote answer sdp: The m= section with mid='0' is invalid. RTCP-MUX is not enabled when it is required. #575

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SAURABHBHAKUNI opened this issue Dec 4, 2024 · 17 comments

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@SAURABHBHAKUNI
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sip.SessionDescriptionHandler | SessionDescriptionHandler.setDescription failed - InvalidAccessError: Failed to execute 'setRemoteDescription' on 'RTCPeerConnection': Failed to set remote answer sdp: The m= section with mid='0' is invalid. RTCP-MUX is not enabled when it is required.

@SAURABHBHAKUNI
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phone.js:2763 Failed to answer call InvalidAccessError: Failed to execute 'setRemoteDescription' on 'RTCPeerConnection': Failed to set remote offer sdp: The m= section with mid='0' is invalid. RTCP-MUX is not enabled when it is required.

@SAURABHBHAKUNI
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iam using asterisk 13 iam ertcp_mux define in sip.conf but same issue.

@InnovateAsterisk
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Can you send the sip trace of the inbound INVITE.

This error is saying that Asterisk is not setting rtcp-mux enabled, but it's required. On the Browser side, you could also try set rtcpMuxPolicy to "negotiate" - this would have to be in the source code, as there isn't a flag for this.

// options.sessionDescriptionHandlerFactoryOptions.peerConnectionConfiguration.rtcpMuxPolicy = "require";

@SAURABHBHAKUNI
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SAURABHBHAKUNI commented Dec 4, 2024 via email

@SAURABHBHAKUNI
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Uploading Screenshot 2024-12-05 014357.png…

@SAURABHBHAKUNI
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geeting the error.

@SAURABHBHAKUNI
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Screenshot 2024-12-05 014357

@SAURABHBHAKUNI
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<--- SIP read from WS:156.67.25.89:61344 --->
INVITE sip:200@myserverip SIP/2.0
Via: SIP/2.0/WSS 192.0.2.158;branch=z9hG4bK3473253
To: sip:200@myserverip
From: "user4" sip:user4@myserverip;tag=h80hfhgg93
CSeq: 1 INVITE
Call-ID: m0hcioiankgj35klpoct
Max-Forwards: 70
Contact: sip:[email protected];transport=wss;ob
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: Browser Phone 0.3.29 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/131.0.0.0 Safari/537.36
Content-Type: application/sdp
Content-Length: 2246

v=0
o=- 7201341312360866036 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS b24b3874-6650-45ae-8a81-83f4461c977f
m=audio 62192 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126
c=IN IP4 45.118.166.200
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:3004991840 1 udp 2122260223 10.8.0.4 62190 typ host generation 0 network-id 3 network-cost 50
a=candidate:4086786002 1 udp 2122194687 192.168.56.1 62191 typ host generation 0 network-id 1
a=candidate:4179499770 1 udp 2122129151 192.168.1.6 62192 typ host generation 0 network-id 2 network-cost 10
a=candidate:3697792272 1 udp 1685921535 45.118.166.200 62192 typ srflx raddr 192.168.1.6 rport 62192 generation 0 network-id 2 network-cost 10
a=candidate:3453185016 1 tcp 1518280447 10.8.0.4 9 typ host tcptype active generation 0 network-id 3 network-cost 50
a=candidate:2371388746 1 tcp 1518214911 192.168.56.1 9 typ host tcptype active generation 0 network-id 1
a=candidate:2278684770 1 tcp 1518149375 192.168.1.6 9 typ host tcptype active generation 0 network-id 2 network-cost 10
a=candidate:2272190244 1 udp 1686052607 156.67.25.89 62190 typ srflx raddr 10.8.0.4 rport 62190 generation 0 network-id 3 network-cost 50
a=ice-ufrag:O3WQ
a=ice-pwd:UeOZrOAqZYBim6pkUc5afzQ4
a=ice-options:trickle
a=fingerprint:sha-256 41:88:1F:1D:B0:02:8C:18:82:78:D7:52:32:E1:C8:68:C3:DD:16:B0:0C:F5:2A:14:C6:84:4F:1E:CA:98:94:CC
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=sendrecv
a=msid:b24b3874-6650-45ae-8a81-83f4461c977f e8b98860-069d-4337-b07b-60769ed11d42
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:126 telephone-event/8000
a=ssrc:2920066302 cname:TrMGT/5JDzsVb1FJ
a=ssrc:2920066302 msid:b24b3874-6650-45ae-8a81-83f4461c977f e8b98860-069d-4337-b07b-60769ed11d42
<------------->
--- (13 headers 45 lines) ---
Using INVITE request as basis request - m0hcioiankgj35klpoct
Found peer 'user4' for 'user4' from 156.67.25.89:61344

<--- Reliably Transmitting (NAT) to 156.67.25.89:61344 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 192.0.2.158;branch=z9hG4bK3473253;received=156.67.25.89;rport=61344
From: "user4" sip:user4@myserverip;tag=h80hfhgg93
To: sip:200@myserverip;tag=as5d3ec69a
Call-ID: m0hcioiankgj35klpoct
CSeq: 1 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="68771488"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'm0hcioiankgj35klpoct' in 20928 ms (Method: INVITE)

<--- SIP read from WS:156.67.25.89:61344 --->
ACK sip:200@myserverip SIP/2.0
Via: SIP/2.0/WSS 192.0.2.158;branch=z9hG4bK3473253
To: sip:200@myserverip;tag=as5d3ec69a
From: "user4" sip:user4@myserverip;tag=h80hfhgg93
Call-ID: m0hcioiankgj35klpoct
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from WS:156.67.25.89:61344 --->
INVITE sip:200@myserverip SIP/2.0
Via: SIP/2.0/WSS 192.0.2.158;branch=z9hG4bK3824609
To: sip:200@myserverip
From: "user4" sip:user4@myserverip;tag=h80hfhgg93
CSeq: 2 INVITE
Call-ID: m0hcioiankgj35klpoct
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username="user4", realm="asterisk", nonce="68771488", uri="sip:200@myserverip", response="e11c67dc9d17b5c294482a306dec2dab"
Contact: sip:[email protected];transport=wss;ob
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: Browser Phone 0.3.29 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/131.0.0.0 Safari/537.36
Content-Type: application/sdp
Content-Length: 2246

v=0
o=- 7201341312360866036 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS b24b3874-6650-45ae-8a81-83f4461c977f
m=audio 62192 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126
c=IN IP4 45.118.166.200
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:3004991840 1 udp 2122260223 10.8.0.4 62190 typ host generation 0 network-id 3 network-cost 50
a=candidate:4086786002 1 udp 2122194687 192.168.56.1 62191 typ host generation 0 network-id 1
a=candidate:4179499770 1 udp 2122129151 192.168.1.6 62192 typ host generation 0 network-id 2 network-cost 10
a=candidate:3697792272 1 udp 1685921535 45.118.166.200 62192 typ srflx raddr 192.168.1.6 rport 62192 generation 0 network-id 2 network-cost 10
a=candidate:3453185016 1 tcp 1518280447 10.8.0.4 9 typ host tcptype active generation 0 network-id 3 network-cost 50
a=candidate:2371388746 1 tcp 1518214911 192.168.56.1 9 typ host tcptype active generation 0 network-id 1
a=candidate:2278684770 1 tcp 1518149375 192.168.1.6 9 typ host tcptype active generation 0 network-id 2 network-cost 10
a=candidate:2272190244 1 udp 1686052607 156.67.25.89 62190 typ srflx raddr 10.8.0.4 rport 62190 generation 0 network-id 3 network-cost 50
a=ice-ufrag:O3WQ
a=ice-pwd:UeOZrOAqZYBim6pkUc5afzQ4
a=ice-options:trickle
a=fingerprint:sha-256 41:88:1F:1D:B0:02:8C:18:82:78:D7:52:32:E1:C8:68:C3:DD:16:B0:0C:F5:2A:14:C6:84:4F:1E:CA:98:94:CC
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=sendrecv
a=msid:b24b3874-6650-45ae-8a81-83f4461c977f e8b98860-069d-4337-b07b-60769ed11d42
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:126 telephone-event/8000
a=ssrc:2920066302 cname:TrMGT/5JDzsVb1FJ
a=ssrc:2920066302 msid:b24b3874-6650-45ae-8a81-83f4461c977f e8b98860-069d-4337-b07b-60769ed11d42
<------------->
--- (14 headers 45 lines) ---
Using INVITE request as basis request - m0hcioiankgj35klpoct
Found peer 'user4' for 'user4' from 156.67.25.89:61344
== DTLS ECDH initialized (automatic), faster PFS enabled
== DTLS ECDH initialized (automatic), faster PFS enabled
== Using SIP VIDEO TOS bits 136
== Using SIP VIDEO CoS mark 6
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 111
Found RTP audio format 63
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 13
Found RTP audio format 110
Found RTP audio format 126
Found audio description format opus for ID 111
Found unknown media description format red for ID 63
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format CN for ID 13
Found unknown media description format telephone-event for ID 110
Found audio description format telephone-event for ID 126
Capabilities: us - (opus|ulaw|vp8), peer - audio=(ulaw|alaw|g722|opus)/video=(nothing)/text=(nothing), combined - (opus|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
> 0x7fe5bc015260 -- Strict RTP learning after remote address set to: 45.118.166.200:62192
Peer audio RTP is at port 45.118.166.200:62192
Peer doesn't provide video
Looking for 200 in from-extensions (domain myserverip)
sip_route_dump: route/path hop: sip:[email protected];transport=wss;ob

<--- Transmitting (NAT) to 156.67.25.89:61344 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS 192.0.2.158;branch=z9hG4bK3824609;received=156.67.25.89;rport=61344
From: "user4" sip:user4@myserverip;tag=h80hfhgg93
To: sip:200@myserverip
Call-ID: m0hcioiankgj35klpoct
CSeq: 2 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:200@myserverip:0;transport=WS
Content-Length: 0

<------------>
-- Executing [200@from-extensions:1] Gosub("SIP/user4-00000003", "dial-extension,s,1,(user5)") in new stack
-- Executing [s@dial-extension:1] NoOp("SIP/user4-00000003", "Calling: user5") in new stack
-- Executing [s@dial-extension:2] Set("SIP/user4-00000003", "JITTERBUFFER(adaptive)=default") in new stack
-- Executing [s@dial-extension:3] Dial("SIP/user4-00000003", "SIP/user5,30") in new stack
== DTLS ECDH initialized (automatic), faster PFS enabled
== DTLS ECDH initialized (automatic), faster PFS enabled
== Using SIP VIDEO TOS bits 136
== Using SIP VIDEO CoS mark 6
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 19620
Adding codec opus to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 156.67.25.89:61489:
INVITE sip:[email protected];transport=wss SIP/2.0
Via: SIP/2.0/WS myserverip:0;branch=z9hG4bK12180a62;rport
Max-Forwards: 70
From: "saurabh" sip:500@myserverip:0;tag=as68407512
To: sip:[email protected];transport=wss
Contact: sip:500@myserverip:0;transport=WS
Call-ID: 504a6ce10cde4f464a49a5bd2c0af70d@myserverip:0
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Wed, 04 Dec 2024 20:47:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1012

v=0
o=root 1220080796 1220080796 IN IP4 myserverip
s=Asterisk PBX certified/13.13-cert9
c=IN IP4 myserverip
t=0 0
m=audio 19620 RTP/SAVPF 107 0 101
a=rtpmap:107 opus/48000/2
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:60
a=ice-ufrag:08a67c8b6098f43477a82b0a59d69b1e
a=ice-pwd:2086d394302f944123226a5b78902125
a=candidate:Hac3f40c 1 UDP 2130706431 10.195.244.12 19620 typ host
a=candidate:Hac1c82b9 1 UDP 2130706431 172.28.130.185 19620 typ host
a=candidate:S673825ea 1 UDP 1694498815 myserverip 19620 typ srflx raddr 10.195.244.12 rport 19620
a=candidate:Hac3f40c 2 UDP 2130706430 10.195.244.12 19621 typ host
a=candidate:Hac1c82b9 2 UDP 2130706430 172.28.130.185 19621 typ host
a=candidate:S673825ea 2 UDP 1694498814 myserverip 19621 typ srflx raddr 10.195.244.12 rport 19621
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 94:54:E3:26:17:20:14:6B:94:71:83:0B:21:6C:24:B3:45:E4:91:EF:24:E7:2F:B6:FA:4D:EA:DA:1B:D5:BC:9A
a=sendrecv


-- Called SIP/user5

<--- SIP read from WS:156.67.25.89:61489 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS myserverip:0;branch=z9hG4bK12180a62;rport
From: "saurabh" sip:500@myserverip:0;tag=as68407512
To: sip:[email protected];transport=wss
CSeq: 102 INVITE
Call-ID: 504a6ce10cde4f464a49a5bd2c0af70d@myserverip:0
Supported: outbound
User-Agent: Browser Phone 0.3.29 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/131.0.0.0 Safari/537.36
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from WS:156.67.25.89:61489 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS myserverip:0;branch=z9hG4bK12180a62;rport
From: "saurabh" sip:500@myserverip:0;tag=as68407512
To: sip:[email protected];transport=wss;tag=eooevkpg5j
CSeq: 102 INVITE
Call-ID: 504a6ce10cde4f464a49a5bd2c0af70d@myserverip:0
Supported: outbound
User-Agent: Browser Phone 0.3.29 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/131.0.0.0 Safari/537.36
Contact: sip:[email protected];transport=wss
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
sip_route_dump: route/path hop: sip:[email protected];transport=wss
-- SIP/user5-00000004 is ringing

<--- Transmitting (NAT) to 156.67.25.89:61344 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WSS 192.0.2.158;branch=z9hG4bK3824609;received=156.67.25.89;rport=61344
From: "user4" sip:user4@myserverip;tag=h80hfhgg93
To: sip:200@myserverip;tag=as693adaf0
Call-ID: m0hcioiankgj35klpoct
CSeq: 2 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:200@myserverip:0;transport=WS
Content-Length: 0

<------------>

<--- SIP read from WS:156.67.25.89:61489 --->
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/WS myserverip:0;branch=z9hG4bK12180a62;rport
From: "saurabh" sip:500@myserverip:0;tag=as68407512
To: sip:[email protected];transport=wss;tag=eooevkpg5j
CSeq: 102 INVITE
Call-ID: 504a6ce10cde4f464a49a5bd2c0af70d@myserverip:0
Supported: outbound
User-Agent: Browser Phone 0.3.29 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/131.0.0.0 Safari/537.36
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (NAT) to 156.67.25.89:61489:
ACK sip:[email protected];transport=wss SIP/2.0
Via: SIP/2.0/WS myserverip:0;branch=z9hG4bK12180a62;rport
Max-Forwards: 70
From: "saurabh" sip:500@myserverip:0;tag=as68407512
To: sip:[email protected];transport=wss;tag=eooevkpg5j
Contact: sip:500@myserverip:0;transport=WS
Call-ID: 504a6ce10cde4f464a49a5bd2c0af70d@myserverip:0
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0


-- SIP/user5-00000004 redirecting info has changed, passing it to SIP/user4-00000003

<--- Transmitting (NAT) to 156.67.25.89:61344 --->
SIP/2.0 181 Call is being forwarded
Via: SIP/2.0/WSS 192.0.2.158;branch=z9hG4bK3824609;received=156.67.25.89;rport=61344
From: "user4" sip:user4@myserverip;tag=h80hfhgg93
To: sip:200@myserverip;tag=as693adaf0
Call-ID: m0hcioiankgj35klpoct
CSeq: 2 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:200@myserverip:0;transport=WS
Content-Length: 0

<------------>
-- SIP/user5-00000004 is busy
Scheduling destruction of SIP dialog '504a6ce10cde4f464a49a5bd2c0af70d@myserverip:0' in 21056 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [s@dial-extension:4] Hangup("SIP/user4-00000003", "") in new stack
== Spawn extension (dial-extension, s, 4) exited non-zero on 'SIP/user4-00000003'
Scheduling destruction of SIP dialog 'm0hcioiankgj35klpoct' in 20928 ms (Method: INVITE)

<--- Reliably Transmitting (NAT) to 156.67.25.89:61344 --->
SIP/2.0 480 Temporarily unavailable
Via: SIP/2.0/WSS 192.0.2.158;branch=z9hG4bK3824609;received=156.67.25.89;rport=61344
From: "user4" sip:user4@myserverip;tag=h80hfhgg93
To: sip:200@myserverip;tag=as693adaf0
Call-ID: m0hcioiankgj35klpoct
CSeq: 2 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>

<--- SIP read from WS:156.67.25.89:61344 --->
ACK sip:200@myserverip SIP/2.0
Via: SIP/2.0/WSS 192.0.2.158;branch=z9hG4bK3824609
To: sip:200@myserverip;tag=as693adaf0
From: "user4" sip:user4@myserverip;tag=h80hfhgg93
Call-ID: m0hcioiankgj35klpoct
CSeq: 2 ACK
Max-Forwards: 70
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Reliably Transmitting (NAT) to 10.5.68.229:5060:
OPTIONS sip:10.5.68.229 SIP/2.0
Via: SIP/2.0/UDP myserverip:5060;branch=z9hG4bK278bea4c;rport
Max-Forwards: 70
From: "unknown" sip:+911140102000@myserverip;tag=as1650137a
To: sip:10.5.68.229
Contact: sip:+911140102000@myserverip:5060
Call-ID: 5be4815f363972a35ddc83cd04672edf@myserverip:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk
Date: Wed, 04 Dec 2024 20:48:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


[Dec 5 02:18:04] ERROR[5777]: chan_sip.c:4269 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data
Reliably Transmitting (NAT) to 156.67.25.89:21346:
OPTIONS sip:[email protected]:21346;rinstance=046cf158e321ddac SIP/2.0
Via: SIP/2.0/UDP myserverip:5060;branch=z9hG4bK7bf076cc;rport
Max-Forwards: 70
From: "unknown" sip:unknown@myserverip;tag=as65fd891f
To: sip:[email protected]:21346;rinstance=046cf158e321ddac
Contact: sip:unknown@myserverip:5060
Call-ID: 78abe2741be3e01447f3d62d1f52f003@myserverip:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk
Date: Wed, 04 Dec 2024 20:48:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


[Dec 5 02:18:04] ERROR[5777]: chan_sip.c:4269 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data
Really destroying SIP dialog '5be4815f363972a35ddc83cd04672edf@myserverip:5060' Method: OPTIONS
Really destroying SIP dialog '78abe2741be3e01447f3d62d1f52f003@myserverip:5060' Method: OPTIONS

@InnovateAsterisk
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Here (from above) the invite does not contain a=rtcp-mux

INVITE sip:[email protected];transport=wss SIP/2.0
Via: SIP/2.0/WS myserverip:0;branch=z9hG4bK12180a62;rport
Max-Forwards: 70
From: "saurabh" sip:500@myserverip:0;tag=as68407512
To: sip:[email protected];transport=wss
Contact: sip:500@myserverip:0;transport=WS
Call-ID: 504a6ce10cde4f464a49a5bd2c0af70d@myserverip:0
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Wed, 04 Dec 2024 20:47:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1012

v=0
o=root 1220080796 1220080796 IN IP4 myserverip
s=Asterisk PBX certified/13.13-cert9
c=IN IP4 myserverip
t=0 0
m=audio 19620 RTP/SAVPF 107 0 101
a=rtpmap:107 opus/48000/2
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:60
a=ice-ufrag:08a67c8b6098f43477a82b0a59d69b1e
a=ice-pwd:2086d394302f944123226a5b78902125
a=candidate:Hac3f40c 1 UDP 2130706431 10.195.244.12 19620 typ host
a=candidate:Hac1c82b9 1 UDP 2130706431 172.28.130.185 19620 typ host
a=candidate:S673825ea 1 UDP 1694498815 myserverip 19620 typ srflx raddr 10.195.244.12 rport 19620
a=candidate:Hac3f40c 2 UDP 2130706430 10.195.244.12 19621 typ host
a=candidate:Hac1c82b9 2 UDP 2130706430 172.28.130.185 19621 typ host
a=candidate:S673825ea 2 UDP 1694498814 myserverip 19621 typ srflx raddr 10.195.244.12 rport 19621
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 94:54:E3:26:17:20:14:6B:94:71:83:0B:21:6C:24:B3:45:E4:91:EF:24:E7:2F:B6:FA:4D:EA:DA:1B:D5:BC:9A
a=sendrecv
``

The asterisk server is not applying the rtcp-mux config 

@SAURABHBHAKUNI
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SAURABHBHAKUNI commented Dec 4, 2024 via email

@SAURABHBHAKUNI
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SAURABHBHAKUNI commented Dec 4, 2024 via email

@InnovateAsterisk
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you could also try set rtcpMuxPolicy to "negotiate" - this would have to be in the source code, as there isn't a flag for this.

or double check the config, for example:

https://github.com/InnovateAsterisk/Browser-Phone/blob/master/config/sip.conf

[webrtc](!)
transport=wss
allow=opus,ulaw,vp9,vp8,h264
encryption=yes
avpf=yes
force_avp=yes
icesupport=yes
rtcp_mux=yes
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/home/pi/certs/raspberrypi.pem
dtlscafile=/home/pi/ca/InnovateAsterisk-Root-CA.crt
dtlssetup=actpass

@InnovateAsterisk
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Its pass the above please chdck previous full log

Yes, this was the invite session from the browser to the server.

User-Agent: Browser Phone 0.3.29 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/131.0.0.0 Safari/537.36

@SAURABHBHAKUNI
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SAURABHBHAKUNI commented Dec 4, 2024 via email

@SAURABHBHAKUNI
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SAURABHBHAKUNI commented Dec 4, 2024 via email

@SAURABHBHAKUNI
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SAURABHBHAKUNI commented Dec 4, 2024 via email

@InnovateAsterisk
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InnovateAsterisk commented Dec 5, 2024

Before you jump into the code, check that Asterisk is setup correctly first.

To be clear the issue as with the B-side endpoint (200)... although 200 DID seems mapped to "User5"

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