[asterisk-dev] UDP/TLS/RTP/SAVPF to RTP/SAVPF

[Lorenzo Miniero]

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 <jaflong at 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
>
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