Re: [Ekiga-list] ekiga registration in asterisk

[Damien Sandras <dsandras seconix com>]

Hi,

Le mardi 04 mars 2008 à 11:35 -0500, sean darcy a écrit :
> Damien Sandras wrote:
> ............
> >> Here's what ekiga.net says it received:
> >> <--- SIP read from UDP://86.64.162.35:5060 --->
> >> SIP/2.0 406 Not Acceptable
> >> Via: SIP/2.0/UDP
> >> 10.10.11.180:5060;branch=z9hG4bK17818198;rport=5060;received=96.xxx.253.yy
> >> .............
> >>
> >> so it knows both the public and the private. Does ekiga require both 
> >> addresses to be public? If so, anybody know how to force asterisk to do 
> >> this?
> > 
> > No, you misunderstood.
> > Ekiga sees it receives a packet from Ekiga.net (86.64.162.35).
> > That packet is the answer from ekiga.net, which contains portions of the
> > original request, which in turn contains only private IP addresses,
> > which is wrong.
> 
> I must be slow here, but if I turn on sip debug in asterisk it shows:

[...]

> I assume the first stanza is the registration packet sent by asterisk to 
> ekiga, and the bottom stanza is the ekiga.net response.
> 
>  From the bottom stanza, it seems ekiga.net "knows" the public ip 
> address - 99.xxx.253.yy ( see the 4th line of the stanza  and the last 
> line), as well as the private address. But this is not enough?

No it is not enough. Because even though we can guess the public IP
address and port for the signalisation messages, they are impossible to
guess when it comes to audio and video RTP traffic. That is the reason
why ekiga.net is configured to reject your REGISTER PDU : because it
contains private IP addresses and ports, and thus it means your system
is not NAT aware and subsequent calls will fail.

> How should the registration packet be changed? According to the asterisk 
> docs, it should meet  RFC3581 (;rport). Does it? If not, then it's a bug 
> in asterisk.
> 
> Or is RFC3581 not enough for ekiga.net? In which case, asterisk needs an 
> additional way of sending sip registration requests to meet ekiga's 
> requirements.

RFC3581 will help for signalisation, however, it won't help for the RTP
streams. You have to configure Asterisk in such a way that the PDUs
contain public IP addresses and ports.

It can be done either by using STUN, or by using directives like
externip + port forwarding for the RTP ports.
-- 
 _     Damien Sandras
(o-      
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