Modify SDP or change Codecs in SDP

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WebRTC

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Aug 3, 2022, 2:53:09 PM8/3/22
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Hello, I need change the SDP, specifically in the m= line by the audio and RTP/AVP. Additionally, i need delete others parameters that JSSIP send in the INVITE. When i modify the SDP in the event 'sdp' from the RTCSession, I received "488 Not Acceptable Here" and the cause print "Bad Media Description".
I modified my SDP as follow:

 session.on('sdp', data => {
 data.sdp =
          'v=0\r\n' +
          'o=WebRTC 1983 678902 IN IP4 181.xxx.xxx.50\r\n' +
          's=WebRTC \r\n' +
          'c=IN IP4 181.xxx.xxx.50\r\n' +
          't=0 0\r\n' +
          'm=audio 57592 RTP/AVP 0 101\r\n' +
          'a=rtpmap:101 telephone-event/8000\r\n' +
          'a=rtpmap:0 PCMU/8000\r\n' +
          'a=ptime:20\r\n' +
          'a=fmtp:101 0-15\r\n' +
          'a=sendrecv\r\n'
}

And I adjunt the log generated.

It should be noted that I am pointing to a server of my provider, therefore I cannot access to modify the configurations of said server

Thanks for you help
logWEBRTC Jssip.txt

Super-Potion

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Oct 5, 2022, 2:05:28 AM10/5/22
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Running into a similar problem. Did you find a fix for this?

WebRTC

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Oct 5, 2022, 10:31:39 AM10/5/22
to JsSIP
Hello, the only solution is to have a gateway like OverSIP, OpenSIP, FreeSwitch, Asterisk, or some SBC that serves as a translator from WebSocket to SIP
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