[SR-Users] Proxying INVITE message without port information in R-URI
Alex Balashov
abalashov at evaristesys.com
Tue Jan 18 19:14:25 CET 2011
Vikram,
1. The port is not mandatory.
2. You can force the port in the RURI explicitly by manipulating the
mutable pseudovariable $rp prior to relay,
E.g.
$rp = 5067;
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
On Jan 18, 2011, at 1:02 PM, Vikram Ragukumar
<vragukumar at signalogic.com> wrote:
> Hello,
>
> The diagram below shows the SIP message flow in our setup when
> Softphone A places a call to Softphone B.
>
> ----------- ----- ---------- ----- -----------
> |Softphone A|-<->-|Proxy|-<->-|Sip Server|-<->-|Proxy|-<->-|
> Softphone B|
> ----------- ----- ---------- ----- -----------
> IP: a.b.c.d IP: a.b.c.d
>
> Proxies shown in the diagram above are physically the same entity.
> For convenience of diagrammatic representation the figure above has
> been unwrapped.
>
> Both Softphone A and Softphone B (hereafter referred to as A,B) are
> behind a NAT and are registered to the Sip Server through the same
> proxy (IP: a.b.c.d). Proxy runs Kamailio v3.0.1.
>
> When A places a call to B, an INVITE is proxied to the Sip Server
> from A. The Sip Server now sends another INVITE to the proxy,
> however the R-URI of this INVITE does not contain the port number
> that the INVITE is supposed to be sent to. Hence the proxy sends the
> INVITE to the default SIP port 5060. This port is not open in the
> firewall hence Softphone B never receives the INVITE.
>
> My questions are
> 1) Is it mandatory for R-URI of the INVITE to contain the port
> number ?
>
> 2) If not, is there something that can be added to the config script
> in Kamailio so that the proxy forwards the INVITE to the appropriate
> port? (maybe by keeping track of the rcv port of the REGISTER
> message that Softphone B sends...) ?
>
> Thanks and Regards,
> Vikram.
>
> PS : The Sip Server in the above discussion is Voipswitch.
>
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