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Unable to RTCPeerConnection::setRemoteDescription: Error Failed to set remote answer sdp: The m= section:audio is invalid. RTCP-MUX is not enabled when it is required. #41
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Please help me how to solve this. Based on this functionality we will decide to choose the platform either flutter or not. |
Please confirm your server supports RTCP-MUX. |
in web side it's working. is there any difference between web and mobile? |
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Please help me out, I am getting same issue. |
This is a webrtc SDP negotiation problem, you need to analyze the offer/answer SDP. |
can you provide me sample code. |
I have used your example with my server details, call connected after 6 sec I got issue like] Bad Media Description, Reason: Not Acceptable Here @cloudwebrtc please help me out |
@cloudwebrtc please update on it. |
This is not the fault of dart-sip-ua itself, but the problem of SDP negotiation. You should analyze the problem from the offer/answer process. You must have a SIP server that supports the WebRTC media transfer protocol, which means that the offer SDP that you send from your SIP Server to dart-sip-ua must follow the WebRTC standard. Because the webrtc native library in dart-sip-ua is compiled by Google. |
We done from web app, it's working good. Our server supported WebRTC. |
@cloudwebrtc can you please provide me your test server, I can test from end. kindly provide me. |
@cloudwebrtc pls update me |
@maheshlalu You can analyze the differences in SDP between normal calls and failed calls, and you should be able to locate the problem quickly. |
@cloudwebrtc I have set the 'rtcpMuxPolicy':'negotiate' in SIPUAHelper class and then |
@maheshlalu You can use Wireshark to capture and analyze udp to confirm that the media stream is connected |
@cloudwebrtc pls look into below log. media not transferred. flutter: [2019-12-31 13:11:35.779] Level.debug config.dart:252 ::: Check optional parameter => user_agent. REGISTER sip:*********01.buniapp.com SIP/2.0 SIP/2.0 401 Unauthorized REGISTER sip:*********01.buniapp.com SIP/2.0 SIP/2.0 200 OK INVITE sip:+91*********@*********01.buniapp.com SIP/2.0 v=0 v=0 SIP/2.0 407 Proxy Authentication Required ACK sip:+91*********@01.buniapp.com SIP/2.0 INVITE sip:+91*********@*********01.buniapp.com SIP/2.0 v=0 v=0 SIP/2.0 100 trying -- your call is important to us SIP/2.0 183 Session Progress v=0 SIP/2.0 200 OK v=0 ACK sip:+91*********@..:5060 SIP/2.0 |
@cloudwebrtc my server team told me that this is NAT issue. Please give me any idea how to solve this. |
Dart-SIP-UA to Asterisk:
Asterisk To Dart-SIP-UA:
Your caller and called party is not on the same network segment, and Asterisk is behind NAT. The candidate provided by Asterisk only has the intranet host, so ice cannot connect.
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@cloudwebrtc Thanks for solution. As per your instruction I have configured the turn server then it's working good. But Call disconnected after 90 sec. Please check once. |
@cloudwebrtc I am getting this error emit "peerconnection:setremotedescriptionfailed" [error:Unable to RTCPeerConnection::setRemoteDescription: Error Failed to set remote offer sdp: The order of m-lines in subsequent offer doesn't match order from previous offer/answer.]<…> |
@cloudwebrtc please check below logs. a=fmtp:111 minptime=10;useinbandfec=1 v=0 SIP/2.0 407 Proxy Authentication Required ACK sip:+@.buniapp.com SIP/2.0 INVITE sip:+@.buniapp.com SIP/2.0 v=0 v=0 SIP/2.0 100 trying -- your call is important to us SIP/2.0 183 Session Progress v=0 SIP/2.0 200 OK v=0 ACK sip:+@..:5060 SIP/2.0 INVITE sip:8040x12y@ym5xusjq6ldr.invalid;transport=ws;ob;alias=14.140.145.146 v=0 SIP/2.0 100 Trying SIP/2.0 488 Not Acceptable Here ACK sip:8040x12y@ym5xusjq6ldr.invalid;transport=ws;ob;alias=14.140.145.146 INVITE sip:8040x12y@ym5xusjq6ldr.invalid;transport=ws;ob;alias=14.140.145.146 v=0 BYE sip:+@..:5060 SIP/2.0 SIP/2.0 481 Call leg/transaction does not exist |
@cloudwebrtc pls update on it |
@maheshlalu Fixed ca3a022 |
How to enable audio streem?
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