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For the two versions, almost everything is identical, except the the browser version uses RTCPeerConnection provided by browser while werift version used the RTCPeerConnection provided by werift.
The browser version works likes a charm.
While the werift version throws a lot of "ssrcReceiver not found" error.
I know that werift may not be used as an out-of-box replacement for the browser webRTC, kindly tell me how to fix this issue. Many thanks!
Update: I have found something:
The offer SDP doesn't have a ssrc line.
And the a=rtcp:48037 line doesn't have a IP address. (not sure this matters)
And werift still tries to use the ssrc which doesn't exist in sdp offer.
I implemented my project using werift-rtp only. I manually implemented the SIP/SDP/RTP communication part.
For those who are interested, please check https://github.com/ringcentral/ringcentral-softphone-ts
In my project, the ssrc value doesn't matter at all. and I am sure for each session there will be only one ssrc at each peer. So I don't need to handle multiple ssrc (like in werift do a mapping from ssrc to ssrcReceiver)
I created a test app here https://github.com/tylerlong/ringcentral-webrtc-test
There are two version:
For the two versions, almost everything is identical, except the the browser version uses
RTCPeerConnection
provided by browser while werift version used theRTCPeerConnection
provided by werift.The browser version works likes a charm.
While the werift version throws a lot of "ssrcReceiver not found" error.
I know that werift may not be used as an out-of-box replacement for the browser webRTC, kindly tell me how to fix this issue. Many thanks!
Update: I have found something:
And werift still tries to use the ssrc which doesn't exist in sdp offer.
Detailed debug logs:
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