[Asterisk-code-review] rtp engine: allocate RTP dynamic payloads per session (testsuite[master])

[Kevin Harwell]

Kevin Harwell has posted comments on this change. ( https://gerrit.asterisk.org/5261 )

Change subject: rtp_engine: allocate RTP dynamic payloads per session
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Patch Set 1:

(1 comment)

https://gerrit.asterisk.org/#/c/5261/1/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/configs/ast1/pjsip.conf
File tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/configs/ast1/pjsip.conf:

Line 3: debug=yes
> Is there a reason you're enabling debug by default on this test?
This test failed initially while testing, so I enabled SIP debug to help. I left it on so if the test fails again in the future more info will be available.

Also currently all tests that use the sipp scenario module enable SIP level debugging. This test bypassed that, but still used sipp in its test, so I figured leaving it enabled brought it more in line with the other tests.


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Gerrit-MessageType: comment
Gerrit-Change-Id: I9070e13981c1c43cc09a7640bc93563df3f18263
Gerrit-PatchSet: 1
Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-Owner: Kevin Harwell <kharwell at digium.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>
Gerrit-Reviewer: Matthew Fredrickson <creslin at digium.com>
Gerrit-HasComments: Yes