[asterisk-bugs] [JIRA] (ASTERISK-29963) res_rtp_asterisk: mapping->ssrc_invalid on unidirectional videostream after confbridge reinvite
Joshua C. Colp (JIRA)
noreply at issues.asterisk.org
Thu Mar 10 05:14:06 CST 2022
[ https://issues.asterisk.org/jira/browse/ASTERISK-29963?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Joshua C. Colp updated ASTERISK-29963:
--------------------------------------
Description:
When a client joins a Confbridge and there is another participant in the conference already then the client get an INVITE from asterisk looking like this (see attached file)
Asterisk will send the other participants video stream on the second video with ssrc=12683906 and everything is working with video and voice coming from and to both participants.
The problem is when the client loses a packet and sends a NACK request to Asterisk, then asterisk is unable to map that rtcp instance to the sendonly stream since the mapping is marked as ssrc_invalid=0. Asterisk will then fail to map to the correct instance and try to act on the voice instance instead and write res_rtp_asterisk.c:6551 ast_rtcp_interpret: (0x7ff0c8083bc0) RTCP before handle NACK request, retransmissions are not enabled ignore this message!
I can see by adding some extra logging in asterisk that the function ast_rtp_bundle sets the mapping.ssrc_valid=child_rtp->themssrc_valid wich in this case is 0 right after asterisk processes the sdp.
It's possible to get retransmissions to work by changing this line in __rtp_find_instance_by_ssrc
if (mapping->ssrc_valid && mapping_ssrc == ssrc) {
to
if (mapping_ssrc == ssrc) {
was:
When a client joins a Confbridge and there is another participant in the conference already then the client get an INVITE from asterisk looking like this
INVITE sip:14hjh1lr at 192.168.1.64:61731;transport=ws;ob SIP/2.0
Via: SIP/2.0/WSS 192.168.1.231:9062;rport;branch=z9hG4bKPjafc6d2ba-2841-43e9-9184-92fb5d27f465;alias
From: <sip:9988 at 192.168.1.231>;tag=f044d3f5-97b1-4305-86e8-d7f6d6f57feb
To: <sip:1_webrtc at 192.168.1.231>;tag=27mgt9i7b4
Contact: <sip:192.168.1.231:9062;transport=ws>
Call-ID: vralkvok6h1vp2r8ab6c
CSeq: 14043 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800;refresher=uas
Min-SE: 90
Max-Forwards: 70
User-Agent: wx3.se pbx
Content-Type: application/sdp
Content-Length: 2599
v=0
o=- 1395058744 5 IN IP4 192.168.1.231
s=Asterisk
c=IN IP4 192.168.1.231
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0 1 video-2
m=audio 14264 UDP/TLS/RTP/SAVPF 111 9 8 0 126
a=connection:existing
a=setup:actpass
a=fingerprint:SHA-256 49:A0:82:79:46:CC:46:F0:92:57:E9:15:5D:B6:11:A1:4E:C4:31:8E:4B:C9:1C:B9:1C:72:DA:23:FD:9E:49:BE
a=ice-ufrag:4c67f5b33099618527fbe21c2244c63a
a=ice-pwd:34a17def72e1fcc62c4a9c990cc02295
a=candidate:Hc0a801e7 1 UDP 2130706431 192.168.1.231 14264 typ host
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:767905177 cname:0e429237-8399-4f4c-ab63-ed627706fffd
a=msid:46167cbd-25d9-46c1-a6d5-80d3cf7cb7a2 70a3431c-0a02-45ac-8890-59adaaa1095a
a=rtcp-fb:* transport-cc
a=mid:0
m=video 14264 UDP/TLS/RTP/SAVPF 98 102 96
a=connection:existing
a=setup:actpass
a=fingerprint:SHA-256 49:A0:82:79:46:CC:46:F0:92:57:E9:15:5D:B6:11:A1:4E:C4:31:8E:4B:C9:1C:B9:1C:72:DA:23:FD:9E:49:BE
a=ice-ufrag:4c67f5b33099618527fbe21c2244c63a
a=ice-pwd:34a17def72e1fcc62c4a9c990cc02295
a=rtpmap:98 VP9/90000
a=rtpmap:102 H264/90000
a=fmtp:102 packetization-mode=1;level-asymmetry-allowed=1;profile-level-id=42001F
a=rtpmap:96 VP8/90000
a=sendrecv
a=rtcp-mux
a=ssrc:1528787399 cname:939a44f3-407e-4b38-a965-8f06ab7fe96c
a=msid:46167cbd-25d9-46c1-a6d5-80d3cf7cb7a2 57cf4b7a-ef1b-4088-88e3-e41ba3cd6034
a=rtcp-fb:* transport-cc
a=rtcp-fb:* ccm fir
a=rtcp-fb:* nack
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=mid:1
m=video 14264 UDP/TLS/RTP/SAVPF 98 102 96
a=connection:existing
a=setup:actpass
a=fingerprint:SHA-256 49:A0:82:79:46:CC:46:F0:92:57:E9:15:5D:B6:11:A1:4E:C4:31:8E:4B:C9:1C:B9:1C:72:DA:23:FD:9E:49:BE
a=ice-ufrag:4c67f5b33099618527fbe21c2244c63a
a=ice-pwd:34a17def72e1fcc62c4a9c990cc02295
a=rtpmap:98 VP9/90000
a=rtpmap:102 H264/90000
a=fmtp:102 packetization-mode=1;level-asymmetry-allowed=1;profile-level-id=42001F
a=rtpmap:96 VP8/90000
a=sendonly
a=rtcp-mux
a=ssrc:733408114 cname:15205e51-0bd9-4d35-a90f-59614506084a
a=msid:dcd91543-5b85-4550-8047-4a585743e859 16b8b8c5-1661-4a1a-8f5d-39f566c1d834
a=rtcp-fb:* transport-cc
a=rtcp-fb:* ccm fir
a=rtcp-fb:* nack
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=mid:video-2
SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.168.1.231:9062;rport;branch=z9hG4bKPjafc6d2ba-2841-43e9-9184-92fb5d27f465;alias
To: <sip:1_webrtc at 192.168.1.231>;tag=27mgt9i7b4
From: <sip:9988 at 192.168.1.231>;tag=f044d3f5-97b1-4305-86e8-d7f6d6f57feb
Call-ID: vralkvok6h1vp2r8ab6c
CSeq: 14043 INVITE
Contact: <sip:14hjh1lr at gn423gqo2vbs.invalid;transport=ws;ob>
Session-Expires: 1800;refresher=uas
Supported: timer,ice,replaces,outbound
Content-Type: application/sdp
Content-Length: 3426
v=0
o=- 849152999030188088 3 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0 1 video-2
a=msid-semantic: WMS u6tstBvLcUbxW0IEheMslXD1t1b6fIKF4cVM
m=audio 60600 UDP/TLS/RTP/SAVPF 111 9 8 0 126
c=IN IP4 212.247.4.149
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:3460887983 1 udp 2122260223 192.168.1.64 60600 typ host generation 0 network-id 1
a=candidate:945327227 1 udp 1686052607 212.247.4.149 60600 typ srflx raddr 192.168.1.64 rport 60600 generation 0 network-id 1
a=candidate:2160789855 1 tcp 1518280447 192.168.1.64 9 typ host tcptype active generation 0 network-id 1
a=ice-ufrag:Nk03
a=ice-pwd:wkJ1TKIZ6kf3QyBGgL6QYZXz
a=ice-options:trickle
a=fingerprint:sha-256 5A:29:86:1F:EE:42:53:0F:14:2C:90:9A:02:19:C3:0E:DE:78:74:D6:2E:15:DD:47:F5:CE:4D:03:72:45:7D:2B
a=setup:passive
a=mid:0
a=sendrecv
a=msid:u6tstBvLcUbxW0IEheMslXD1t1b6fIKF4cVM b8a51bfa-2ff0-4c5a-8b5e-0f3374fa0c66
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=ssrc:1854519133 cname:LDd7LE4FD0g2MJyG
m=video 63435 UDP/TLS/RTP/SAVPF 98 102 96
c=IN IP4 212.247.4.149
b=AS:1000a=rtcp:9 IN IP4 0.0.0.0
a=candidate:3460887983 1 udp 2122260223 192.168.1.64 63435 typ host generation 0 network-id 1
a=candidate:945327227 1 udp 1686052607 212.247.4.149 63435 typ srflx raddr 192.168.1.64 rport 63435 generation 0 network-id 1
a=candidate:2160789855 1 tcp 1518280447 192.168.1.64 9 typ host tcptype active generation 0 network-id 1
a=ice-ufrag:Nk03
a=ice-pwd:wkJ1TKIZ6kf3QyBGgL6QYZXz
a=ice-options:trickle
a=fingerprint:sha-256 5A:29:86:1F:EE:42:53:0F:14:2C:90:9A:02:19:C3:0E:DE:78:74:D6:2E:15:DD:47:F5:CE:4D:03:72:45:7D:2B
a=setup:passive
a=mid:1
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=sendrecv
a=msid:u6tstBvLcUbxW0IEheMslXD1t1b6fIKF4cVM d1f0a022-7164-49a7-acb9-74e9ef8bbee6
a=rtcp-mux
a=rtpmap:98 VP9/90000
a=rtcp-fb:98 transport-cc
a=rtcp-fb:98 ccm fir
a=rtcp-fb:98 nack
a=fmtp:98 profile-id=0
a=rtpmap:102 H264/90000
a=rtcp-fb:102 transport-cc
a=rtcp-fb:102 ccm fir
a=rtcp-fb:102 nack
a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 transport-cc
a=rtcp-fb:96 ccm fir
a=rtcp-fb:96 nack
a=ssrc-group:FID 12683906 4121512
a=ssrc:12683906 cname:LDd7LE4FD0g2MJyG
a=ssrc:4121512 cname:LDd7LE4FD0g2MJyG
m=video 9 UDP/TLS/RTP/SAVPF 98 102 96
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:Nk03
a=ice-pwd:wkJ1TKIZ6kf3QyBGgL6QYZXz
a=ice-options:trickle
a=fingerprint:sha-256 5A:29:86:1F:EE:42:53:0F:14:2C:90:9A:02:19:C3:0E:DE:78:74:D6:2E:15:DD:47:F5:CE:4D:03:72:45:7D:2B
a=setup:passive
a=mid:video-2
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=recvonly
a=rtcp-mux
a=rtpmap:98 VP9/90000
a=rtcp-fb:98 transport-cc
a=rtcp-fb:98 ccm fir
a=rtcp-fb:98 nack
a=fmtp:98 profile-id=0
a=rtpmap:102 H264/90000
a=rtcp-fb:102 transport-cc
a=rtcp-fb:102 ccm fir
a=rtcp-fb:102 nack
a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 transport-cc
a=rtcp-fb:96 ccm fir
a=rtcp-fb:96 nack
Asterisk will send the other participants video stream on the second video with ssrc=12683906 and everything is working with video and voice coming from and to both participants.
The problem is when the client loses a packet and sends a NACK request to Asterisk, then asterisk is unable to map that rtcp instance to the sendonly stream since the mapping is marked as ssrc_invalid=0. Asterisk will then fail to map to the correct instance and try to act on the voice instance instead and write res_rtp_asterisk.c:6551 ast_rtcp_interpret: (0x7ff0c8083bc0) RTCP before handle NACK request, retransmissions are not enabled ignore this message!
I can see by adding some extra logging in asterisk that the function ast_rtp_bundle sets the mapping.ssrc_valid=child_rtp->themssrc_valid wich in this case is 0 right after asterisk processes the sdp.
It's possible to get retransmissions to work by changing this line in __rtp_find_instance_by_ssrc
if (mapping->ssrc_valid && mapping_ssrc == ssrc) {
to
if (mapping_ssrc == ssrc) {
> res_rtp_asterisk: mapping->ssrc_invalid on unidirectional videostream after confbridge reinvite
> -----------------------------------------------------------------------------------------------
>
> Key: ASTERISK-29963
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-29963
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Resources/res_rtp_asterisk
> Affects Versions: 18.8.0
> Environment: Alma Linux 8, Chrome WebRTC with JsSIP
> Reporter: Erik Bergschöld
> Labels: webrtc
> Attachments: sip.txt
>
>
> When a client joins a Confbridge and there is another participant in the conference already then the client get an INVITE from asterisk looking like this (see attached file)
> Asterisk will send the other participants video stream on the second video with ssrc=12683906 and everything is working with video and voice coming from and to both participants.
> The problem is when the client loses a packet and sends a NACK request to Asterisk, then asterisk is unable to map that rtcp instance to the sendonly stream since the mapping is marked as ssrc_invalid=0. Asterisk will then fail to map to the correct instance and try to act on the voice instance instead and write res_rtp_asterisk.c:6551 ast_rtcp_interpret: (0x7ff0c8083bc0) RTCP before handle NACK request, retransmissions are not enabled ignore this message!
> I can see by adding some extra logging in asterisk that the function ast_rtp_bundle sets the mapping.ssrc_valid=child_rtp->themssrc_valid wich in this case is 0 right after asterisk processes the sdp.
> It's possible to get retransmissions to work by changing this line in __rtp_find_instance_by_ssrc
> if (mapping->ssrc_valid && mapping_ssrc == ssrc) {
> to
> if (mapping_ssrc == ssrc) {
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