[Kamailio-Users] stun/outbound draft...

[Aymeric Moizard]

On Fri, 2 Jan 2009, Iñaki Baz Castillo wrote:

> 2009/1/2 Aymeric Moizard <jack at atosc.org>:
>
> I have used the ser_stun.c/.h files to add STUN support
> on the socket of kamalio.
>
> I have modified the XOR-MAPPED-ADDRESS results because
> wireshark is not analysing it the same way although I'm
> not very sure who is right on this: wireshark versus ser.
>
> I hope you guys can work on merging and testing. Coming
> from ser, I guess the code is pretty good!


Let me a question. What is the purpose/advantage of having a STUN
server running in the SIP proxy/registrar port?
AFAIK this is just valid for a single purpose: allowing STUN *just for
SIP signalling* when the device is located behind symmetric NAT (in
which "normal" STUN doesn't work since the public mapping the router
assigns depends also on the destination ip:port, and not just on the
private source ip:port).

This is: having a STUN server running in port 5060 in the same host
where our SIP proxy/registrar runs is just valid for a UA behind
symmetric NAT because it can set the "Contact" header with the mapped
public ip:port, so it will be able to receive in-dialog requests
(without NAT solution at SIP level in the proxy), but it will never be
valid for RTP, since the destination of the RTP will never be
PROXY_IP:5060, so the mapping our symmetric NAT router will do for the
RTP is completely unknown.

So, what is the advantage of a STUN server in port 5060? Thanks.


    This specification draft-ietf-sip-outbound-16 ios only about SIP:
    http://tools.ietf.org/html/draft-ietf-sip-outbound-16

    So for 100% of UA, STUN is usable for keeping alive and detecting
    IP changes on the SIP connection.

    Don't you want kamailio to be standard?

    For RTP? Even if you put a STUN server on another port, that would
    not be of any help at all... because only part of the 100% can use
    the STUN discovered address in their RTP and such solution would not
    work 100%.
    Instead for RTP, ICE and TURN are required. That's just a different
    server that has to be installed: NOT THE SAME as the one running on
    the SIP server which is ONLY doing STUN binding request.


    tks,
    Aymeric MOIZARD / ANTISIP
    amsip - http://www.antisip.com
    osip2 - http://www.osip.org
    eXosip2 - http://savannah.nongnu.org/projects/exosip/



-- 
Iñaki Baz Castillo
<ibc at aliax.net>
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