pespin has submitted this change. ( https://gerrit.osmocom.org/c/docker-playground/+/36951?usp=email )

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(-)

Approvals: Jenkins Builder: Verified laforge: Looks good to me, approved

diff --git a/ttcn3-asterisk-ims-ue-test/asterisk/pjsip.conf b/ttcn3-asterisk-ims-ue-test/asterisk/pjsip.conf index 9400780..9ca85e5 100644 --- a/ttcn3-asterisk-ims-ue-test/asterisk/pjsip.conf +++ b/ttcn3-asterisk-ims-ue-test/asterisk/pjsip.conf @@ -106,6 +106,7 @@ rewrite_contact=yes from_user=238010000090828 from_domain=172.18.11.103 +direct_media=no

[volte_ims] type=auth