[Spce-user] sipwise 3.1 (CE) and websocket support

[Andrew Pogrebennyk]

On 14/02/14 01:33, Ashutosh Apte wrote:
> I've also created a SIP Peering Group to route calls to a legacy SIP
> gateway. On dialing, the call hits the legacy gateway but I notice that
> the SDP body in the INVITE message uses RTP/SAVF. This gateway does not
> support sRTP. Is there a way to force the mediaproxy to use RTP only?
> 
> I've changed the parameter under Domains -> Preferences -> NAT and Media
> Flow Control -> srtp_transcoding from "transparent" to "ForceRTP". The
> kamailio proxy debug logs do show that the parameter was indeed changed
> but it did not have any effect on the INVITE message.
> 
> What configuration parameter should I be changing to get this to work?

Hi,
from my PoV you should set srtp_transcoding in the domain to "Prefer
SRTP" (AFAIK it doesn't work if you leave it as "transparent" and I want
to add automatic detection of endpoints that use RTP/SAVP and RTP/SAVPF)
and most important - srtp_transcoding in the peer preferences (for your
legacy gateway) to "Force RTP". This will enable transcoding. You may
also need to set rtcp_feedback to "Prefer AVPF" on the domain and "Force
AVP" on the legacy gateway.

Andrew