Attention is currently required from: Hoernchen, jolly, laforge.
Hello Hoernchen, Jenkins Builder, jolly, laforge,
I'd like you to reexamine a change. Please visit
https://gerrit.osmocom.org/c/docker-playground/+/36951?usp=email
to look at the new patch set (#2).
Change subject: asterisk: pjsip.conf: Disable remote bridging between local SIP and IMS
......................................................................
asterisk: pjsip.conf: Disable remote bridging between local SIP and IMS
While implementing a first ttcn3 test validating the MO call scenario
(SIP-UA -> Asterisk -> IMS-CORE) [1] I was running into the scenario
where, after the first SIP INVITE + 200 OK + ACK, Asterisk was sending
a RE-INVITE to both parties to attempt to remotely bridge them (RTP
traffic flowing directly between them without passing through Asterisk).
This happened in part because I'm so far configure A-LAW on both sides
so asterisk figures out it can do so. I still need to change IMS-core
to EVS only.
In any case, regardless of the codecs used, my understanding is that
asterisk should never attempt remote bridging when using the "volte_ims"
endpoint, since that network segment is separate from the local network
where the local SIP UAs are located.
The "direct_media=no" option just addresses the issue; Asterisk no longer
tries to re-invite after the call is established
Related: SYS#6782
Change-Id: I4edea96151b31f02bf292b43b757922389375429
---
M ttcn3-asterisk-ims-ue-test/asterisk/pjsip.conf
1 file changed, 29 insertions(+), 0 deletions(-)
git pull ssh://gerrit.osmocom.org:29418/docker-playground refs/changes/51/36951/2
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Gerrit-Project: docker-playground
Gerrit-Branch: master
Gerrit-Change-Id: I4edea96151b31f02bf292b43b757922389375429
Gerrit-Change-Number: 36951
Gerrit-PatchSet: 2
Gerrit-Owner: pespin <pespin(a)sysmocom.de>
Gerrit-Reviewer: Hoernchen <ewild(a)sysmocom.de>
Gerrit-Reviewer: Jenkins Builder
Gerrit-Reviewer: jolly <andreas(a)eversberg.eu>
Gerrit-Reviewer: laforge <laforge(a)osmocom.org>
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