[asterisk-dev] s?rtp via SIP/TLS/TCP

[Olle E. Johansson]

On 21 Apr 2015, at 17:55, James Cloos <cloos at jhcloos.com> wrote:

>>>>>> "OEJ" == Olle E Johansson <oej at edvina.net> writes:
> 
> OEJ> It's a bug in chan_sip that I fixed a while ago in one of my branches.
> OEJ> SNOM sends an SDES key but RTP/AVP in the offer and chan_sip
> OEJ> chokes. It's a one or two line fix - or turn off SRTP in the SNOM.
> 
> I presume this one?:
> 
> sdes-rtp-avp.diff:
> Index: channels/chan_sip.c
> ===================================================================
> --- channels/chan_sip.c	(revision 417744)
> +++ channels/chan_sip.c	(working copy)
> @@ -9603,7 +9603,7 @@
> 				if (audio) {
> 					if (process_sdp_a_sendonly(value, &sendonly)) {
> 						processed = TRUE;
> -					} else if (!processed_crypto && process_crypto(p, p->rtp, &p->srtp, value)) {
> +					} else if (secure_audio && !processed_crypto && process_crypto(p, p->rtp, &p->srtp, value)) {
> 						processed_crypto = TRUE;
> 						processed = TRUE;
> 					} else if (process_sdp_a_audio(value, p, &newaudiortp, &last_rtpmap_codec)) {
> 
> But, as I wrote, it still failed w/o rtp crypto and forcing avpf works.

Would be interesting to see the INVITE.

/O