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authorPau Espin Pedrol <pespin@sysmocom.de>2024-05-30 13:37:09 +0200
committerPau Espin Pedrol <pespin@sysmocom.de>2024-05-30 13:41:01 +0200
commitc9a787ad1fde5f70beed8e763d6b12ccc2cd4a4b (patch)
tree9f2f2de1e582ef45b28354ae9c4702d519e99633 /osmo-bts-latest/Dockerfile
parentcc5ec183c0f4414116a9628025a1395655a67fa6 (diff)
asterisk: pjsip.conf: Disable remote bridging between local SIP and IMSHEADmaster
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
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