[Freeswitch-users] INVITE via another SIP proxy

[Ingmar Schraub]

Meanwhile we could debug this problem and found that the proxy didn't do
the content-length calculation correctly in one scenario. Interestingly
FS / Sofia-SIP is here much more strict than any other sip-stack/vendor
which includes the list of vendors I have mentioned previously!

So FS is great and I really appreciate the fact that it does follow the
RFCs as strict as possible! I wish other vendors would be as strict
because this would sort out quite some stupid problems right from the
beginning!

No further action needed :-)

Thanks a lot,

Ingmar

On 11/25/2010 12:00 AM, Michael Collins wrote:
> Definitely pastebin the console log with siptrace. We'll take a look.
> Most likely it is working with Asterisk, Avaya, 3Com, Nortel, Cisco, and
> Sonus is because they all take liberties with the SIP specs. FreeSWITCH
> actually attempts to follow the RFCs, at least wherever they aren't
> horribly incoherent or just plain stupid. :)
> 
> Put the log on pastebin.freeswitch.org <http://pastebin.freeswitch.org>
> and paste the link in this thread.
> -MC
> 
> On Mon, Nov 22, 2010 at 9:23 AM, Ingmar Schraub <is at eseco.de
> <mailto:is at eseco.de>> wrote:
> 
>     Hi,
> 
>     I ran into a problem today and I thought I should place the question
>     here and hopefully someone has an idea what's going on.
> 
>     FYI, I am using GIT from November 15th 2010.
> 
>     Here is my problem:
> 
>     My SIP phones are all on the LAN. I can register them, make calls, etc.
>     all work fine. I run FS with "-nonat".
> 
>     Now, when I add another SIP proxy in the chain, let my phones register
>     with FS via that proxy, everything is still fine. No issues with
>     registrations.
> 
>     The problem is, that INVITEs do not work. FS rejects them with "400 Bad
>     Request".
> 
>     I turned on SIP tracing on FS to see how the SIP messages look like and
>     I noticed that FS is for some reason stripping off (apparently..) the
>     SDP part from the INVITE.
> 
>     Why do I believe that? Well, I used tcpdump on the FS box to capture the
>     network traffic and see what arrives actually. The INVITE received by
>     the network stack shows the full INVITE message including the SDP part.
> 
>     On the other hand, when I make a call / INVITE from a phone which is
>     directly registered at FS, the INVITE message shown in the sip trace
>     includes the SDP.
> 
>     Thus, I believe that FS has an issue parsing that particular INVITE
>     message properly.
> 
>     Note: the same proxy, the same set-up works all sorts of other PBXs and
>     Soft-Switches (Asterisk, Avaya, 3com, Nortel, Cisco, Sonus, ...). It's
>     all tested.
> 
>     Of course I can provide the FS SIP trace log and the corresponding
>     tcpdump. If any developer could have a quick look, I'd really appreciate
>     this.
> 
>     Thanks and best regards,
> 
>     Ingmar
> 
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> 
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