WebRTC calls - wrong SDP

Well, I had a really long investigation here (https://github.com/versatica/JsSIP/issues/46) and the result is - Asterisk doesn’t react correctly on some SDP parts. Could you guys please check?

Both Chrome and Firefox set “RTP/SAVPF” on m line in SDP and Asterisk doesn’t apply DTLS :frowning: (Asterisk expectations are “UDP/TLS/RTP/SAVPF”)

All the logs and details are here https://github.com/versatica/JsSIP/issues/46

Asterisk 11.7.

Howdy,

The forums aren’t a great place for bringing this up, as most of the people watching these forums aren’t deeply involved with developing Asterisk. You probably want to raise this concern on the asterisk-dev mailing list. lists.digium.com/mailman/listinfo/asterisk-dev

[quote=“malcolmd”]Howdy,

The forums aren’t a great place for bringing this up, as most of the people watching these forums aren’t deeply involved with developing Asterisk. You probably want to raise this concern on the asterisk-dev mailing list. lists.digium.com/mailman/listinfo/asterisk-dev[/quote]

Ok, thanks. Did as you say.
BTW - more people complain about DTLS + Asterisk, because Firefox uses only DTLS - https://github.com/versatica/JsSIP/issues/189

[quote=“iskomorokh”][quote=“malcolmd”]Howdy,

The forums aren’t a great place for bringing this up, as most of the people watching these forums aren’t deeply involved with developing Asterisk. You probably want to raise this concern on the asterisk-dev mailing list. lists.digium.com/mailman/listinfo/asterisk-dev[/quote]

Ok, thanks. Did as you say.
BTW - more people complain about DTLS + Asterisk, because Firefox uses only DTLS - https://github.com/versatica/JsSIP/issues/189[/quote]
Here have my opinion, Webrtc is not a full designed protocol, there is no release yet all is kind of hype, so based on that and in the developer side did you think that worth to waste hours implementing such protocol? We have the users have an awful experience with the Google voice because that.

Said that if you are releasing a product based on webrtc you are making the wrong move or too early and finally keep in mind that the folks of jssip doesn’t have an intention to create the API for work smoothly with asterisk since in their words make an API to connect only to PBX for PSTN is just silly.