This has been discussed several times, actually. Asterisk does the right thing, because the standard mandates UDP/TLS/RTP/SAVPF when DTLS is involved. I don't know why browsers only use RTP/SAVPF instead. That said, it's something that you can easily fix directly in JavaScript.
Lorenzo 2014-04-04 13:14 GMT+02:00, jaflong jaflong <jafl...@yandex.com>: > > Hi List, > > Can anyone please advise where sdp m = line can be modified in the source > code (chan_sip.c) > > for example the I want to change m=audio 30490 UDP/TLS/RTP/SAVPF 0 126 to > m=audio 30490 RTP/SAVPF 0 126 > > UDP/TLS/RTP/SAVPF to RTP/SAVPF > > on the response invite > > where in the code can this be done > > > Regards > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
