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Re: [Partysip-dev] Call setup problems


From: Aymeric Moizard
Subject: Re: [Partysip-dev] Call setup problems
Date: Wed, 23 Apr 2003 17:39:53 +0200 (CEST)


On Wed, 23 Apr 2003, Marian Durkovic wrote:

> Hi all,
>
>
>    I'm trying to setup patrysip 0.5.6. as a proxy server for SIP calls.
> However, I've run into the following problems:
>
> 1) when trying to make a call, the phone rings only for 30 seconds. After
> this amount of time partysip sends CANCEL to the called party and
> 408 Request Timeout to the calling party. It does not accept the Expires
> header in INVITE message, which requests e.g. 180 seconds timeout.

This is already fixed in the soon to come partysip-0.5.7.
You can grab the CVS which includes the required modifications.

Note that, 0.5.7 will not look at the expires header...

> 2) partysip drops the ACK after 200 OK message in case the client sends it
> via partysip. Yes, there's the following comment in src/psp_core3.c:
>
>  /* It can be a ACK for 200 ok, but those ACK SHOULD
>  never go through this proxy! (and should be sent to the contact header */
>
> But there are applications which do it this way

I don't like them :)

> - for example sip-h323
> translators, which only talk to the nearest sip proxy and thus are sending
> the ACKs after 200 OK to them as well. Also some UAs do it like this.
> When partysip drops the ACK, the call could not be established.

The rfc does not allow this behavior and mandates to use the
record-routing mechanism.

There is actually one way to deal with this issue: configure partysip to
do "record-routing". This way evry SIP messages will go through the proxy
and partysip will handle them correctly. (because of the presence of
the Route header). (Note that MSN reply with an error when receiving
calls from a proxy that does record-routing and which conforms to
the latest rfc... It's not partysip's fault again!)

To enable record-routing, you have to modify **evrywhere** in the config
file lines. (it appears more than once)

  record-route off
in
  record-route on

Aymeric

>       Thanks in advance for your help.
>
>
> --------------------------------------------------------------------------
> ----                                                                  ----
> ----   Marian Durkovic                       network  manager         ----
> ----                                                                  ----
> ----   Slovak Technical University           Tel: +421 2 524 51 301   ----
> ----   Computer Centre, Nam. Slobody 17      Fax: +421 2 524 94 351   ----
> ----   812 43 Bratislava, Slovak Republic    E-mail: address@hidden        
> ----
> ----                                                                  ----
> --------------------------------------------------------------------------
>
>
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