[asterisk-dev] [Code Review] 2796: Testsuite: sip_hold_direct_media adaptations for Asterisk 12

[jrose]

-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2796/
-----------------------------------------------------------

(Updated Aug. 28, 2013, 4:37 p.m.)


Review request for Asterisk Developers, Joshua Colp, kmoore, Matt Jordan, and Mark Michelson.


Changes
-------

Fix the directmedia aspect of the test for Asterisk 11, unify the SIPP scenarios.

It turned out that we weren't sending SDP data in 200 OK responses when we really should have been for this test. Asterisk 11 has a check in place to revert to local bridging when this is the case while Asterisk 12 doesn't, but the conclusion Mark and I came to was that really this behavior is inappropriate for both versions. The fallback behavior difference is a consequence of the new native RTP bridging code not containing an address family check that the old RTP engine bridging code used when deciding on local or remote bridging, but a fix should probably be centered on checking that 200 OKs in invite transactions contain SDP.


Bugs: https://issues.asterisk.org/jira/browse/ASTERISK-22217
    https://issues.asterisk.org/jira/browse/https://issues.asterisk.org/jira/browse/ASTERISK-22217


Repository: testsuite


Description
-------

On Friday I committed a patch which addressed some bugs with holding in Asterisk 12 while using native RTP bridges and directmedia. As part of that effort, the SIP hold tests in the testsuite were split up and divided into tests which used direct media and tests which didn't use direct media. At the time, Asterisk 12 failed the tests which used direct media. This patch fixes those tests by making the test use Asterisk 12 specific sipp scenarios (which were based on the existing scenarios). The main difference between the Asterisk 12 scenarios and their older counterparts was always just the addition of more expected invites and responses to those invites on account of how directmedia is established in Asterisk 12 both during the initial setup.


Diffs (updated)
-----

  /asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_B_media_restrict.xml 4089 
  /asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_B_IP_restrict.xml 4089 
  /asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_B_IP_media_restrict.xml 4089 
  /asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_A_IP_restrict.xml PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/run-test 4089 
  /asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/inject_bypass.csv 4089 
  /asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_A.xml 4089 

Diff: https://reviewboard.asterisk.org/r/2796/diff/


Testing
-------

Ran modified tests against Asterisk 12.
Ran modified tests against Asterisk 11.

Repeated this process many times to make sure the results were consistent.
Tracked the invites in 12 against the code that was generating them.


Thanks,

jrose

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20130828/7a02ddbd/attachment-0001.htm>