On Mon, Apr 7, 2014 at 3:41 PM, Olle E. Johansson <oej@edvina.net> wrote:

On 07 Apr 2014, at 15:39, Andreas Granig <agranig@sipwise.com> wrote:

> Hi,
>
> On 03/26/2014 03:18 PM, Alex Balashov wrote:
>> No. You can't route the RTP and RTCP traffic to Kamailio, by definition.
>>
>> You keep asking questions that betray a lack of basic understanding of SIP network elements. I think you should take Olle's suggestion and learn how it works.
>
> For the sake of discussion, I think it's somewhat possible to route
> rtp/rtcp with kamailio. Does it make sense? No. Would it work? Probably,
> in a limited way.

Of course, if you are a developer, you can do anything. :-)

But the question wasn't asked in the developer context, at least I did not
parse it that way...

/O

>
> So if I wanted to do something like this, then I'd find the point where
> kamailio is actually calling recv(), then find out where it feeds the
> received data into the sip parser. There, I'd implement the logic to
> quickly check if what we're dealing with is an rtp packet, and handle it
> differently than other packets. For SDP in request and response bodies
> flowing through my config, I'd modify SDP to put 5060 as media port for
> the various streams.
>
> Now since every packet will be received on port 5060, you can't really
> distinguish between different streams, as you can't rely on the source
> address advertised in SDP because of NAT, so any NAT scenario with more
> than one phone behind that NAT is going to break the whole thing. Well,
> putting aside NAT, you now would have to maintain mapping tables of
> source addresses announced in SDP and check (and rely on) them for
> inbound packets and map them to the outbound leg based on the source
> address. That might work for non-NAT scenarios (but who's using NAT in a
> world of IPv6 anyways?).
>
> Now the question is, why would anyone want to do that? If the intention
> is to make it work better in NAT environments, then our OP has probably
> not thought it through entirely.
>
> Andreas
>
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