4 Jun
2012
4 Jun
'12
1:37 p.m.
Hi,
just an idea:
Why not look at MGCP or Megaco/H.248 to control the RTP-Proxy? Might
be interesting, since you could use Kamailio as a Client for other
cases, such as Announcement-Servers or PSTN Termination....
This needs more investigation, i know, and it's way more complicated,
but maybe worth the effort; i don't know.
Just an idea.
Carsten
2012/6/2 Andreas Granig <agranig(a)sipwise.com>om>:
Hi Peter,
On 06/01/2012 07:25 PM, Peter Lemenkov wrote:
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My personal opinion is that we should pass the
entire SDP (wrapped
into json perhaps) - codec mapping and their parameters, encryption
keys, stun data - this all could be useful for the rtpproxy backend.
Otherwise it looks good.
When taking into account all kinds of use-cases like ICE handling
between ICE/non-ICE clients and srtp bridging for webrtc, as well as
transcoding for "normal" clients, it makes sense to pass the full SDP
body to the rtpproxy and expecting a full SDP body back in response.
Maybe some basic wrapping which includes generic and SIP specific values
(like the cookie, call-id, tags, branches, IPv4/IPv6 bridging etc) as
well as the full SDP for a request, and for responses a status and again
the full SDP would be sufficient.
That way, we'd get also get rid of certain flags in the rtpproxy module.
Andreas
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