asterisk users - Apr 2013 - On SIP INVITE answering to IP:port found in Contact: header.

Hi list!

I'm trying to get a DID routed to me and the provider seems to have an 
unusual setup. Or maybe not? From looking at their SIP header they are 
using "BroadWorks".

The problem: they're sending their SIP invite from port 36252. My 
Asterisk 10.7.1 is answering to that port 36252 but their BroadWorks 
thingie is not listening on that port, but instead on port 5060. So they 
want me to send my SIP responses to the IP:port they send via the 
Contact: header.

Here's the output from tcpdump (1.1.1.1 = them; 2.2.2.2 = me):

IP 1.1.1.1.36252 > 2.2.2.2.5060: UDP, length 845
[...]
Contact:<sip:1.1.1.1:5060>
[...]

My Asterisk' reply:

IP 2.2.2.2.5060 > 1.1.1.1.36252: UDP, length 602
[...]

I tried the following in the peer config in sip.conf:

host=1.1.1.1
type=peer
insecure=port,invite
port=5060

But that didn't change anything.

How can I tell Asterisk to honor and use the info found in Contact: when 
sending its replies?

Thank you!
Markus

Markus,

I think I know what's wrong here but I did a fair amount of research while
digging into your problem.  I may have misinterpreted something along the way so
you should also consider other responses, especially if they come from someone
who claims greater expertise.  I did this to help you and to learn, so I would
appreciate your feedback and welcome any corrections from more knowledgeable
list members.

Now on with the show.  My responses are inline.
 
> I'm trying to get a DID routed to me and the provider seems to have an 
> unusual setup. Or maybe not? From looking at their SIP header they are 
> using "BroadWorks".
Using BroadWorks isn't unusual, but the rest of what you described is.
 
> The problem: they're sending their SIP invite from port 36252. My 
> Asterisk 10.7.1 is answering to that port 36252 but their BroadWorks 
> thingie is not listening on that port, but instead on port 5060. So they 
> want me to send my SIP responses to the IP:port they send via the 
> Contact: header.
That's where it got weird.  From RFC 3261 [1]:

   While the Via header field tells other elements where to send the
   response, the Contact header field tells other elements where to send
   future requests.
   ...
   The Contact header field provides a SIP or SIPS URI that can be used
   to contact that specific instance of the UA for subsequent requests.
   ...
   The Via header field indicates the transport used for the transaction
   and identifies the location where the response is to be sent.
   ...
   A Via header field value contains the transport protocol used to send
   the message, the client's host name or network address, and possibly
   the port number at which it wishes to receive responses.
> Here's the output from tcpdump (1.1.1.1 = them; 2.2.2.2 = me):
> 
> IP 1.1.1.1.36252 > 2.2.2.2.5060: UDP, length 845
> [...]
> Contact:<sip:1.1.1.1:5060>
> [...]
> 
> My Asterisk' reply:
> 
> IP 2.2.2.2.5060 > 1.1.1.1.36252: UDP, length 602
> [...]
Their INVITES are requests and your Asterisk's replies (e.g. 100 Trying, 183
Session Progress, 200 OK) are responses.  As the RFC specifies, the Contact
header field applies to future requests.  However, it is the Via header field
that is used to route responses [2][3].
> I tried the following in the peer config in sip.conf:
> 
> host=1.1.1.1
> type=peer
> insecure=port,invite
> port=5060
> 
> But that didn't change anything.
> 
> How can I tell Asterisk to honor and use the info found in Contact: when 
> sending its replies?
I don't think that you can, because Asterisk is probably just following the
specification and sending responses back to whatever IP and port are in the Via
header of the INVITE.  It's hard to say with certainty without seeing the
entire
SIP dialog, but I think the provider needs to modify their INVITE to have a Via
header like the following:

  Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK2e91efaf

[1] http://www.ietf.org/rfc/rfc3261.txt
[2]
https://lists.cs.columbia.edu/pipermail/sip-implementors/2005-December/011517.html
[3] http://www.sipknowledge.com/faq_child5.htm#q2.12

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

Markus wrote:
> Hi list!
>
> I'm trying to get a DID routed to me and the provider seems to have an
> unusual setup. Or maybe not? From looking at their SIP header they are
> using "BroadWorks".
>
> The problem: they're sending their SIP invite from port 36252. My
> Asterisk 10.7.1 is answering to that port 36252 but their BroadWorks
> thingie is not listening on that port, but instead on port 5060. So they
> want me to send my SIP responses to the IP:port they send via the
> Contact: header.
If you set nat=no for that specific peer it should work as you need. 
'rport' is forced on these days which works for most situations, except 
with some platforms and Cisco phones. >_>

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org