[Asterisk-code-review] res pjsip: Add direct media rtp keepalive test (testsuite[master])

[Mark Michelson]

Mark Michelson has uploaded a new patch set (#2).

Change subject: res_pjsip: Add direct media rtp_keepalive test
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res_pjsip: Add direct media rtp_keepalive test

This test has two SIPp scenarios bridged together in Asterisk with
direct media enabled. The test ensures that when direct media is in use,
Asterisk does not send any RTP keepalive packets to the endpoints.

ASTERISK-25242
Reported by Mark Michelson

Change-Id: I23009970201cc07701f8208a652b682f64c11537
---
A tests/channels/pjsip/rtp/rtp_keepalive/direct_media/configs/ast1/extensions.conf
A tests/channels/pjsip/rtp/rtp_keepalive/direct_media/configs/ast1/pjsip.conf
A tests/channels/pjsip/rtp/rtp_keepalive/direct_media/rtp.py
A tests/channels/pjsip/rtp/rtp_keepalive/direct_media/sipp/alice.xml
A tests/channels/pjsip/rtp/rtp_keepalive/direct_media/sipp/bob.xml
A tests/channels/pjsip/rtp/rtp_keepalive/direct_media/test-config.yaml
M tests/channels/pjsip/rtp/rtp_keepalive/tests.yaml
7 files changed, 336 insertions(+), 0 deletions(-)


  git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/00/900/2
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Gerrit-MessageType: newpatchset
Gerrit-Change-Id: I23009970201cc07701f8208a652b682f64c11537
Gerrit-PatchSet: 2
Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-Owner: Mark Michelson <mmichelson at digium.com>
Gerrit-Reviewer: Matt Jordan <mjordan at digium.com>