[asterisk-dev] [Code Review] Test that outboundproxy is used for followup authentication INVITEs

[Matt Jordan]

-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/1992/#review6505
-----------------------------------------------------------

Ship it!


Hooray for tests!

- Matt


On June 19, 2012, 10 a.m., Mark Michelson wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/1992/
> -----------------------------------------------------------
> 
> (Updated June 19, 2012, 10 a.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> This scenario sets up three SIP peers.
> One is a UAC (Alice), bound to port 5063. And two are UASes, bound to port 5061 (Carol) and 5065 (Bob). We also have set outboundproxy to be the localhost, port 5062 (Percy).
> 
> We run three SIPp scenarios. One corresponds to Carol, one corresponds to Percy, and one corresponds to Alice. Alice attempts to call Bob. Those of you who are observant will notice that we do not have a SIPp scenario running for Bob.
> 
> Here is the expected call flow
> 
> Alice             *                     Percy              Carol
> ---INVITE (Bob)-->
>                    ---INVITE (Bob)------->
>                    <-401 (Contact: Carol)-
>                    --------ACK----------->
>                    ---INVITE (Bob)------->
>                    <-200 (Contact: Carol)-
>                    -------------------------ACK-------------->
> <-----200 OK------
> ------ACK-------->
> ------BYE-------->
> <-----200 OK------
>                    -------------------------BYE-------------->
>                    <---------------------200 OK---------------
> 
> The idea is that the outboundproxy should be the destination for the initial INVITE, despite the fact that Bob is the target. Then, despite the fact that Carol is specified as the Contact in the 401 to Asterisk, the ensuing ACK and INVITE with authentication still should go to Percy. Once Percy sends a 200 OK with Carol's contact, however, then further SIP traffic should be routed to Carol.
> 
> I've linked in ASTERISK-20008 because without the patch attached to that issue, this test fails.
> 
> 
> This addresses bug ASTERISK-20008.
>     https://issues.asterisk.org/jira/browse/ASTERISK-20008
> 
> 
> Diffs
> -----
> 
>   /asterisk/trunk/tests/channels/SIP/sip_outbound_proxy/configs/ast1/extensions.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_outbound_proxy/configs/ast1/sip.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_outbound_proxy/run-test PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_outbound_proxy/sipp/uac.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_outbound_proxy/sipp/uas-ackonly.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_outbound_proxy/sipp/uas.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_outbound_proxy/test-config.yaml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/tests.yaml 3263 
> 
> Diff: https://reviewboard.asterisk.org/r/1992/diff
> 
> 
> Testing
> -------
> 
> 
> Thanks,
> 
> Mark
> 
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20120619/cfbf37a6/attachment.htm>