Failed to execute 'setRemoteDescription' on 'RTCPeerConnection': Failed to set remote answer sdpWebRTC muxing using BUNDLE in JavaDoes Kurento-utils WebRtcPeerSendrecv pay attention to all options?App RTC calls not working from ios to iosWhat's missing in Answer SDP (From web browser to android device)Failed to set remote video description send parameters on native IOSWebrtc SDP Audio is not working on sender side but receiver can listen sender voiceFailed to set remote video description send parameters IPhone / AndroidComunicate Webrtc Android to Webrtc C# not workRemote offer sdp: Failed to set remote video description send parametersInvalidAccessError: Failed to set remote offer sdp: Failed to set remote video description send parameters
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Failed to execute 'setRemoteDescription' on 'RTCPeerConnection': Failed to set remote answer sdp
WebRTC muxing using BUNDLE in JavaDoes Kurento-utils WebRtcPeerSendrecv pay attention to all options?App RTC calls not working from ios to iosWhat's missing in Answer SDP (From web browser to android device)Failed to set remote video description send parameters on native IOSWebrtc SDP Audio is not working on sender side but receiver can listen sender voiceFailed to set remote video description send parameters IPhone / AndroidComunicate Webrtc Android to Webrtc C# not workRemote offer sdp: Failed to set remote video description send parametersInvalidAccessError: Failed to set remote offer sdp: Failed to set remote video description send parameters
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I have implemented WebRTC for video-audio call through browser. I am using the latest adapter.js. I also implemented WebRTC for native android app too. My signaling server is based on php WebSocket(Ratchet websocket). When I'm testing my implementation within two browsers(chrome-chrome, or chrome-firefox combination) I see the protocol is working fine. I mean video-audio call goes properly. When I test my android app I can see it's also working for app to app communication.
But the problem is when I'm trying to give call from my android app to web app I see the setRemoteDescription is not working rather it's generating the following error:
Failed to execute 'setRemoteDescription' on 'RTCPeerConnection': Failed to set remote answer sdp: Media section has more than one track specified with a=ssrc lines which is not supported with Unified Plan.
In my gradle(app) script I implemented the WebRTC library as follows:
implementation 'org.webrtc:google-webrtc:1.0.+'
That's why I think the latest adapter.js(for web part) and android library should be compatible and there are no issue related with version incompatibility between the web implementation and android implementation.
For better understanding I am sharing here the SDP message which is found when generating offer.
v=0
o=- 7452034467634633423 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0 1
a=msid-semantic: WMS 90 BOLNAOPqVHAtxWW2PZJyA5RG9IkH2MntC2EP
m=audio 9 UDP/TLS/RTP/SAVPF 111 103 9 0 8 105 13 110 113 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:KwHG
a=ice-pwd:YhH0S06F5dJrCns9jFbscFMA
a=ice-options:trickle renomination
a=fingerprint:sha-256 B1:75:DB:D5:20:8C:86:F1:CC:54:4A:1C:C6:C9:AD:D3:79:C4:1E:45:57:CD:9B:FC:CC:1D:01:C5:C1:C5:BF:93
a=setup:active
a=mid:0
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:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:2787512374 cname:dF4uE3SebLpSsKGh
a=ssrc:2787512374 msid:90 101
a=ssrc:2787512374 mslabel:90
a=ssrc:2787512374 label:101
a=ssrc:4286803808 cname:dF4uE3SebLpSsKGh
a=ssrc:4286803808 msid:BOLNAOPqVHAtxWW2PZJyA5RG9IkH2MntC2EP cf5d6c28-239b-4c64-8051-33dbbb81edf4
a=ssrc:4286803808 mslabel:BOLNAOPqVHAtxWW2PZJyA5RG9IkH2MntC2EP
a=ssrc:4286803808 label:cf5d6c28-239b-4c64-8051-33dbbb81edf4
m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 108 109 124
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:KwHG
a=ice-pwd:YhH0S06F5dJrCns9jFbscFMA
a=ice-options:trickle renomination
a=fingerprint:sha-256 B1:75:DB:D5:20:8C:86:F1:CC:54:4A:1C:C6:C9:AD:D3:79:C4:1E:45:57:CD:9B:FC:CC:1D:01:C5:C1:C5:BF:93
a=setup:active
a=mid:1
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:4 urn:3gpp:video-orientation
a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=extmap:10 http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07
a=sendrecv
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 goog-remb
a=rtcp-fb:96 transport-cc
a=rtcp-fb:96 ccm fir
a=rtcp-fb:96 nack
a=rtcp-fb:96 nack pli
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
a=rtpmap:98 VP9/90000
a=rtcp-fb:98 goog-remb
a=rtcp-fb:98 transport-cc
a=rtcp-fb:98 ccm fir
a=rtcp-fb:98 nack
a=rtcp-fb:98 nack pli
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=98
a=rtpmap:108 red/90000
a=rtpmap:109 rtx/90000
a=fmtp:109 apt=108
a=rtpmap:124 ulpfec/90000
a=ssrc-group:FID 1166964128 3959308542
a=ssrc:1166964128 cname:dF4uE3SebLpSsKGh
a=ssrc:1166964128 msid:90 100
a=ssrc:1166964128 mslabel:90
a=ssrc:1166964128 label:100
a=ssrc:3959308542 cname:dF4uE3SebLpSsKGh
a=ssrc:3959308542 msid:90 100
a=ssrc:3959308542 mslabel:90
a=ssrc:3959308542 label:100
a=ssrc-group:FID 1617372799 1848430645
a=ssrc:1617372799 cname:dF4uE3SebLpSsKGh
a=ssrc:1617372799 msid:BOLNAOPqVHAtxWW2PZJyA5RG9IkH2MntC2EP b4a261f3-065b-47bd-a759-207401cb9a6e
a=ssrc:1617372799 mslabel:BOLNAOPqVHAtxWW2PZJyA5RG9IkH2MntC2EP
a=ssrc:1617372799 label:b4a261f3-065b-47bd-a759-207401cb9a6e
a=ssrc:1848430645 cname:dF4uE3SebLpSsKGh
a=ssrc:1848430645 msid:BOLNAOPqVHAtxWW2PZJyA5RG9IkH2MntC2EP b4a261f3-065b-47bd-a759-207401cb9a6e
a=ssrc:1848430645 mslabel:BOLNAOPqVHAtxWW2PZJyA5RG9IkH2MntC2EP
a=ssrc:1848430645 label:b4a261f3-065b-47bd-a759-207401cb9a6e
javascript
android webrtc phpwebsocket asked Mar 27 at 13:56
Mushfiqur RahmanMushfiqur Rahman
9011 bronze badges
add a comment |
I have implemented WebRTC for video-audio call through browser. I am using the latest adapter.js. I also implemented WebRTC for native android app too. My signaling server is based on php WebSocket(Ratchet websocket). When I'm testing my implementation within two browsers(chrome-chrome, or chrome-firefox combination) I see the protocol is working fine. I mean video-audio call goes properly. When I test my android app I can see it's also working for app to app communication.
But the problem is when I'm trying to give call from my android app to web app I see the setRemoteDescription is not working rather it's generating the following error:
Failed to execute 'setRemoteDescription' on 'RTCPeerConnection': Failed to set remote answer sdp: Media section has more than one track specified with a=ssrc lines which is not supported with Unified Plan.
In my gradle(app) script I implemented the WebRTC library as follows:
implementation 'org.webrtc:google-webrtc:1.0.+'
That's why I think the latest adapter.js(for web part) and android library should be compatible and there are no issue related with version incompatibility between the web implementation and android implementation.
For better understanding I am sharing here the SDP message which is found when generating offer.
v=0
o=- 7452034467634633423 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0 1
a=msid-semantic: WMS 90 BOLNAOPqVHAtxWW2PZJyA5RG9IkH2MntC2EP
m=audio 9 UDP/TLS/RTP/SAVPF 111 103 9 0 8 105 13 110 113 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:KwHG
a=ice-pwd:YhH0S06F5dJrCns9jFbscFMA
a=ice-options:trickle renomination
a=fingerprint:sha-256 B1:75:DB:D5:20:8C:86:F1:CC:54:4A:1C:C6:C9:AD:D3:79:C4:1E:45:57:CD:9B:FC:CC:1D:01:C5:C1:C5:BF:93
a=setup:active
a=mid:0
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:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:2787512374 cname:dF4uE3SebLpSsKGh
a=ssrc:2787512374 msid:90 101
a=ssrc:2787512374 mslabel:90
a=ssrc:2787512374 label:101
a=ssrc:4286803808 cname:dF4uE3SebLpSsKGh
a=ssrc:4286803808 msid:BOLNAOPqVHAtxWW2PZJyA5RG9IkH2MntC2EP cf5d6c28-239b-4c64-8051-33dbbb81edf4
a=ssrc:4286803808 mslabel:BOLNAOPqVHAtxWW2PZJyA5RG9IkH2MntC2EP
a=ssrc:4286803808 label:cf5d6c28-239b-4c64-8051-33dbbb81edf4
m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 108 109 124
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:KwHG
a=ice-pwd:YhH0S06F5dJrCns9jFbscFMA
a=ice-options:trickle renomination
a=fingerprint:sha-256 B1:75:DB:D5:20:8C:86:F1:CC:54:4A:1C:C6:C9:AD:D3:79:C4:1E:45:57:CD:9B:FC:CC:1D:01:C5:C1:C5:BF:93
a=setup:active
a=mid:1
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:4 urn:3gpp:video-orientation
a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=extmap:10 http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07
a=sendrecv
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 goog-remb
a=rtcp-fb:96 transport-cc
a=rtcp-fb:96 ccm fir
a=rtcp-fb:96 nack
a=rtcp-fb:96 nack pli
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
a=rtpmap:98 VP9/90000
a=rtcp-fb:98 goog-remb
a=rtcp-fb:98 transport-cc
a=rtcp-fb:98 ccm fir
a=rtcp-fb:98 nack
a=rtcp-fb:98 nack pli
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=98
a=rtpmap:108 red/90000
a=rtpmap:109 rtx/90000
a=fmtp:109 apt=108
a=rtpmap:124 ulpfec/90000
a=ssrc-group:FID 1166964128 3959308542
a=ssrc:1166964128 cname:dF4uE3SebLpSsKGh
a=ssrc:1166964128 msid:90 100
a=ssrc:1166964128 mslabel:90
a=ssrc:1166964128 label:100
a=ssrc:3959308542 cname:dF4uE3SebLpSsKGh
a=ssrc:3959308542 msid:90 100
a=ssrc:3959308542 mslabel:90
a=ssrc:3959308542 label:100
a=ssrc-group:FID 1617372799 1848430645
a=ssrc:1617372799 cname:dF4uE3SebLpSsKGh
a=ssrc:1617372799 msid:BOLNAOPqVHAtxWW2PZJyA5RG9IkH2MntC2EP b4a261f3-065b-47bd-a759-207401cb9a6e
a=ssrc:1617372799 mslabel:BOLNAOPqVHAtxWW2PZJyA5RG9IkH2MntC2EP
a=ssrc:1617372799 label:b4a261f3-065b-47bd-a759-207401cb9a6e
a=ssrc:1848430645 cname:dF4uE3SebLpSsKGh
a=ssrc:1848430645 msid:BOLNAOPqVHAtxWW2PZJyA5RG9IkH2MntC2EP b4a261f3-065b-47bd-a759-207401cb9a6e
a=ssrc:1848430645 mslabel:BOLNAOPqVHAtxWW2PZJyA5RG9IkH2MntC2EP
a=ssrc:1848430645 label:b4a261f3-065b-47bd-a759-207401cb9a6e
javascript
android webrtc phpwebsocket asked Mar 27 at 13:56
Mushfiqur RahmanMushfiqur Rahman
9011 bronze badges
add a comment |
I have implemented WebRTC for video-audio call through browser. I am using the latest adapter.js. I also implemented WebRTC for native android app too. My signaling server is based on php WebSocket(Ratchet websocket). When I'm testing my implementation within two browsers(chrome-chrome, or chrome-firefox combination) I see the protocol is working fine. I mean video-audio call goes properly. When I test my android app I can see it's also working for app to app communication.
But the problem is when I'm trying to give call from my android app to web app I see the setRemoteDescription is not working rather it's generating the following error:
Failed to execute 'setRemoteDescription' on 'RTCPeerConnection': Failed to set remote answer sdp: Media section has more than one track specified with a=ssrc lines which is not supported with Unified Plan.
In my gradle(app) script I implemented the WebRTC library as follows:
implementation 'org.webrtc:google-webrtc:1.0.+'
That's why I think the latest adapter.js(for web part) and android library should be compatible and there are no issue related with version incompatibility between the web implementation and android implementation.
For better understanding I am sharing here the SDP message which is found when generating offer.
v=0
o=- 7452034467634633423 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0 1
a=msid-semantic: WMS 90 BOLNAOPqVHAtxWW2PZJyA5RG9IkH2MntC2EP
m=audio 9 UDP/TLS/RTP/SAVPF 111 103 9 0 8 105 13 110 113 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:KwHG
a=ice-pwd:YhH0S06F5dJrCns9jFbscFMA
a=ice-options:trickle renomination
a=fingerprint:sha-256 B1:75:DB:D5:20:8C:86:F1:CC:54:4A:1C:C6:C9:AD:D3:79:C4:1E:45:57:CD:9B:FC:CC:1D:01:C5:C1:C5:BF:93
a=setup:active
a=mid:0
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:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:2787512374 cname:dF4uE3SebLpSsKGh
a=ssrc:2787512374 msid:90 101
a=ssrc:2787512374 mslabel:90
a=ssrc:2787512374 label:101
a=ssrc:4286803808 cname:dF4uE3SebLpSsKGh
a=ssrc:4286803808 msid:BOLNAOPqVHAtxWW2PZJyA5RG9IkH2MntC2EP cf5d6c28-239b-4c64-8051-33dbbb81edf4
a=ssrc:4286803808 mslabel:BOLNAOPqVHAtxWW2PZJyA5RG9IkH2MntC2EP
a=ssrc:4286803808 label:cf5d6c28-239b-4c64-8051-33dbbb81edf4
m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 108 109 124
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:KwHG
a=ice-pwd:YhH0S06F5dJrCns9jFbscFMA
a=ice-options:trickle renomination
a=fingerprint:sha-256 B1:75:DB:D5:20:8C:86:F1:CC:54:4A:1C:C6:C9:AD:D3:79:C4:1E:45:57:CD:9B:FC:CC:1D:01:C5:C1:C5:BF:93
a=setup:active
a=mid:1
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:4 urn:3gpp:video-orientation
a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=extmap:10 http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07
a=sendrecv
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 goog-remb
a=rtcp-fb:96 transport-cc
a=rtcp-fb:96 ccm fir
a=rtcp-fb:96 nack
a=rtcp-fb:96 nack pli
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
a=rtpmap:98 VP9/90000
a=rtcp-fb:98 goog-remb
a=rtcp-fb:98 transport-cc
a=rtcp-fb:98 ccm fir
a=rtcp-fb:98 nack
a=rtcp-fb:98 nack pli
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=98
a=rtpmap:108 red/90000
a=rtpmap:109 rtx/90000
a=fmtp:109 apt=108
a=rtpmap:124 ulpfec/90000
a=ssrc-group:FID 1166964128 3959308542
a=ssrc:1166964128 cname:dF4uE3SebLpSsKGh
a=ssrc:1166964128 msid:90 100
a=ssrc:1166964128 mslabel:90
a=ssrc:1166964128 label:100
a=ssrc:3959308542 cname:dF4uE3SebLpSsKGh
a=ssrc:3959308542 msid:90 100
a=ssrc:3959308542 mslabel:90
a=ssrc:3959308542 label:100
a=ssrc-group:FID 1617372799 1848430645
a=ssrc:1617372799 cname:dF4uE3SebLpSsKGh
a=ssrc:1617372799 msid:BOLNAOPqVHAtxWW2PZJyA5RG9IkH2MntC2EP b4a261f3-065b-47bd-a759-207401cb9a6e
a=ssrc:1617372799 mslabel:BOLNAOPqVHAtxWW2PZJyA5RG9IkH2MntC2EP
a=ssrc:1617372799 label:b4a261f3-065b-47bd-a759-207401cb9a6e
a=ssrc:1848430645 cname:dF4uE3SebLpSsKGh
a=ssrc:1848430645 msid:BOLNAOPqVHAtxWW2PZJyA5RG9IkH2MntC2EP b4a261f3-065b-47bd-a759-207401cb9a6e
a=ssrc:1848430645 mslabel:BOLNAOPqVHAtxWW2PZJyA5RG9IkH2MntC2EP
a=ssrc:1848430645 label:b4a261f3-065b-47bd-a759-207401cb9a6e
javascript
android webrtc phpwebsocket asked Mar 27 at 13:56
Mushfiqur RahmanMushfiqur Rahman
9011 bronze badges
I have implemented WebRTC for video-audio call through browser. I am using the latest adapter.js. I also implemented WebRTC for native android app too. My signaling server is based on php WebSocket(Ratchet websocket). When I'm testing my implementation within two browsers(chrome-chrome, or chrome-firefox combination) I see the protocol is working fine. I mean video-audio call goes properly. When I test my android app I can see it's also working for app to app communication.
But the problem is when I'm trying to give call from my android app to web app I see the setRemoteDescription is not working rather it's generating the following error:
Failed to execute 'setRemoteDescription' on 'RTCPeerConnection': Failed to set remote answer sdp: Media section has more than one track specified with a=ssrc lines which is not supported with Unified Plan.
In my gradle(app) script I implemented the WebRTC library as follows:
implementation 'org.webrtc:google-webrtc:1.0.+'
That's why I think the latest adapter.js(for web part) and android library should be compatible and there are no issue related with version incompatibility between the web implementation and android implementation.
For better understanding I am sharing here the SDP message which is found when generating offer.
v=0
o=- 7452034467634633423 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0 1
a=msid-semantic: WMS 90 BOLNAOPqVHAtxWW2PZJyA5RG9IkH2MntC2EP
m=audio 9 UDP/TLS/RTP/SAVPF 111 103 9 0 8 105 13 110 113 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:KwHG
a=ice-pwd:YhH0S06F5dJrCns9jFbscFMA
a=ice-options:trickle renomination
a=fingerprint:sha-256 B1:75:DB:D5:20:8C:86:F1:CC:54:4A:1C:C6:C9:AD:D3:79:C4:1E:45:57:CD:9B:FC:CC:1D:01:C5:C1:C5:BF:93
a=setup:active
a=mid:0
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:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:2787512374 cname:dF4uE3SebLpSsKGh
a=ssrc:2787512374 msid:90 101
a=ssrc:2787512374 mslabel:90
a=ssrc:2787512374 label:101
a=ssrc:4286803808 cname:dF4uE3SebLpSsKGh
a=ssrc:4286803808 msid:BOLNAOPqVHAtxWW2PZJyA5RG9IkH2MntC2EP cf5d6c28-239b-4c64-8051-33dbbb81edf4
a=ssrc:4286803808 mslabel:BOLNAOPqVHAtxWW2PZJyA5RG9IkH2MntC2EP
a=ssrc:4286803808 label:cf5d6c28-239b-4c64-8051-33dbbb81edf4
m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 108 109 124
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:KwHG
a=ice-pwd:YhH0S06F5dJrCns9jFbscFMA
a=ice-options:trickle renomination
a=fingerprint:sha-256 B1:75:DB:D5:20:8C:86:F1:CC:54:4A:1C:C6:C9:AD:D3:79:C4:1E:45:57:CD:9B:FC:CC:1D:01:C5:C1:C5:BF:93
a=setup:active
a=mid:1
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:4 urn:3gpp:video-orientation
a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=extmap:10 http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07
a=sendrecv
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 goog-remb
a=rtcp-fb:96 transport-cc
a=rtcp-fb:96 ccm fir
a=rtcp-fb:96 nack
a=rtcp-fb:96 nack pli
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
a=rtpmap:98 VP9/90000
a=rtcp-fb:98 goog-remb
a=rtcp-fb:98 transport-cc
a=rtcp-fb:98 ccm fir
a=rtcp-fb:98 nack
a=rtcp-fb:98 nack pli
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=98
a=rtpmap:108 red/90000
a=rtpmap:109 rtx/90000
a=fmtp:109 apt=108
a=rtpmap:124 ulpfec/90000
a=ssrc-group:FID 1166964128 3959308542
a=ssrc:1166964128 cname:dF4uE3SebLpSsKGh
a=ssrc:1166964128 msid:90 100
a=ssrc:1166964128 mslabel:90
a=ssrc:1166964128 label:100
a=ssrc:3959308542 cname:dF4uE3SebLpSsKGh
a=ssrc:3959308542 msid:90 100
a=ssrc:3959308542 mslabel:90
a=ssrc:3959308542 label:100
a=ssrc-group:FID 1617372799 1848430645
a=ssrc:1617372799 cname:dF4uE3SebLpSsKGh
a=ssrc:1617372799 msid:BOLNAOPqVHAtxWW2PZJyA5RG9IkH2MntC2EP b4a261f3-065b-47bd-a759-207401cb9a6e
a=ssrc:1617372799 mslabel:BOLNAOPqVHAtxWW2PZJyA5RG9IkH2MntC2EP
a=ssrc:1617372799 label:b4a261f3-065b-47bd-a759-207401cb9a6e
a=ssrc:1848430645 cname:dF4uE3SebLpSsKGh
a=ssrc:1848430645 msid:BOLNAOPqVHAtxWW2PZJyA5RG9IkH2MntC2EP b4a261f3-065b-47bd-a759-207401cb9a6e
a=ssrc:1848430645 mslabel:BOLNAOPqVHAtxWW2PZJyA5RG9IkH2MntC2EP
a=ssrc:1848430645 label:b4a261f3-065b-47bd-a759-207401cb9a6e
javascript
android webrtc phpwebsocket javascript
android webrtc phpwebsocket asked Mar 27 at 13:56
Mushfiqur RahmanMushfiqur Rahman
9011 bronze badges
asked Mar 27 at 13:56
Mushfiqur RahmanMushfiqur Rahman
9011 bronze badges
edited Mar 27 at 14:03
Mushfiqur Rahman
asked Mar 27 at 13:56
Mushfiqur RahmanMushfiqur Rahman
9011 bronze badges
asked Mar 27 at 13:56
Mushfiqur RahmanMushfiqur Rahman
9011 bronze badges
asked Mar 27 at 13:56
Mushfiqur RahmanMushfiqur Rahman
9011 bronze badges
9011 bronze badges
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