Reputation: 1
I use the library JsSIP to make SIP calls over WebRTC plataform in Google Chrome web browser. I have to change the SDP directive "UDP/TLS/RTP/SAVPF" in SIP request to "UDP/RTP/AVPF" in JsSIP.js file because the Asterisk server reject calls no encrypted in TLS context and i need the calls no encrypted. But even with the change, the browser continues to send the "UDP/TLS/RTP/SAVPF" in SIP request. Anyone can help me? The WebRTC implementation or Chrome intercepts the SIP request and change to "UDP/TLS/RTP/SAVPF"? Or in some way, this directive is in cache request and needs a clean up? (I already tried to clean the Chrome cache). Or have another explanation to that?
Upvotes: 0
Views: 2286
Reputation: 1665
Chrome doesn't allow unencrypted/unsecure RTP. It always forces SRTP/DTLS encryption.
Upvotes: 0
Reputation: 2864
You should be able to capture SDP under 'sdp' event thrown by jssip and over write with new sdp
'sdp': function(evt){
evt.sdp = your changed sdp
}
Upvotes: 0