[sr-dev] Redesign of rtpproxy control protocol

[Carsten Bock]

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 at sipwise.com>:
> Hi Peter,
>
> On 06/01/2012 07:25 PM, Peter Lemenkov wrote:
>> 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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>



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