[Serusers] record_route and re-invite problem

[bert berlin]

Below is a call flow for a situation that fails -
I have attached the ser.cfg file, and two ethereal sniffs - these are 
for fax call attempts - one with record_route commented out (which 
succeeds) , and one with record_route(), which fails. 
Note that in the failure case, with record_route(), the re-invite from 
208.226.140.40 gets to the proxy at 208.226.140.141, but the proxy never 
sends that to the origination UA, 208.226.140.142.

The fax case looks nearly identical to the call flow below, which was 
taken from a call attempt with a SIP phone placed on hold.

 From TRT --
We are having a problem with a Re-Invite between and endpoint and the 
SER proxy. The call flow is:

 

 

Assumptions: Record route turned on in Proxy

 

 

UA-Orig                        
                            Proxy                               
         UA-Term

192.168.1.10                                           
192.168.1.20                              192.168.1.30

 

------------------Invite sip:1 at 192.168.1.20------->

<--------------- 100 Trying  ----------------------------

                                                              -------- 
Invite sip:1 at 192.168.1.30 --------->

                                                              
<-------------- 100 Trying --------------------

                                                              
<------------------180 Ringing ----------------

<---------------- 180 Ringing -------------------------

                                                              
<------------------ 200 OK ---------------------

<----------------  200 OK-------------------------------

--------------------- ACK sip:1 at 192.168.1.20 à

                                                              --------- 
Ack sip:1 at 192.168.1.30 ------>

 

******** Call is connected and voice path is established  
*************************

******** Termination puts call on hold and sends NULL SDP via Re-Invite 
******

 

                                                             <-------- 
Invite sip:1 at 192.168.1.20-------

** Note: to and from tags are set correctly, call id is same as original 
Invite, Sequence incremented

                                                               
---------- 100 Trying ------------------------->

                          
                                     ---------- Invite 
sip:1 at 192.168.1.30 -------->

** Invite is being sent back to termination. It is not identifying this 
call as an existing call in the proxy.

 

 

 

TRT
-- 

Bert Berlin

Director, System Test
Quintum Technologies, Inc.
71 James Way
Eatontown,NJ 07724
ph 732-460-9000 ext 247


-------------- next part --------------
An embedded and charset-unspecified text was scrubbed...
Name: ser.cfg.record-route
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20040706/7e3d380b/attachment.txt>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: sip-fax-no-record-route-7-6-04
Type: application/octet-stream
Size: 26039 bytes
Desc: not available
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20040706/7e3d380b/attachment.obj>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: sip-fax-with-record-route-7-6-04
Type: application/octet-stream
Size: 27796 bytes
Desc: not available
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20040706/7e3d380b/attachment-0001.obj>