Comments (4)
Caching may be involved in Sofia (the SIP stack used in the plugin), as I do remember some similar issue when failing an authenticated registration as well.
Are you using redirects for the purpose? Can you share the SIP interaction (without the SDP not to clog the issue) highlighting what's going wrong?
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Okay, I have changed a few things in the logs to protect my server:
XXX.XXX.XXX.XXX => server ip
XXXX-YYYYYYYYYYYYY => SIP account
YYYY-XXXXXXXXXXXXXX => Janus SIP Plugin user
ZZZZZZZZZZZ => incoming context
XXXX => extension that is called from Janus SIP Plugin, this is the same as the XXXX in the SIP account.
SIP log: https://gist.github.com/anonymous/48f43105b02bf3d26548
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Sorry if this took so long...
I've just been able to reproduce the issue. You're right, the plugin is not sending the INVITE to the extension that is passed to it but the actual username of the peer. Looking at the SIP messages, I don't see the extension anywhere, though, which makes me think it's not any caching that is involved (there would be no mapping to make), but a bug in the plugin itself. In fact, the original INVITE has the username as part of the calling address (e.g., sip:caller@ip:5060), which is what is called by the plugin later on. Most likely there's a variable that is not cleaned as it should, which results in the plugin calling the last caller instead of who's requested.
I'll look into it right away.
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It should be fixednow. It was not an issue with variables and the like, but a wrong reuse of libsofia's nua_handles. I just pushed the commit, let me know if that fixes it for you as well and I'll close the issue.
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Related Issues (20)
- [1.x] Sip session get stuck HOT 12
- No call back triggering in 'onlocalstream' HOT 2
- [1.x] Streaming plugin does not send "started" event to event handlers HOT 2
- Add unauthenticated no-payload ping endpoint for health checks HOT 3
- Build fails with latest libsrtp version HOT 9
- [0.x] [janus.js] black screen or streaming stops periodically when connected through WebRTC (Chromium only) HOT 4
- [1.x] Issue with outbound calls for SIP Gateway HOT 3
- [1.x] videoroom: remote publisher doesn't release RTP/RTCP ports HOT 10
- [1.x] Does AudioBridge supports multiple participant join from RTP in a single room . Getting `Already in a room` Error for 2nd participant HOT 3
- [1.x] rtp_port_range not being respected HOT 2
- [1.x] Crash / segfault on 1.2.2 HOT 5
- [1.x] system under heavy load stops to handle calls wia ws api HOT 11
- [1.x] Crash / segfault on 1.2.3 HOT 3
- [1.x] inconsistency happen when more than 6 users join at same time HOT 2
- [1.x]inconsistency happen when more than 6 users join at same time with different browsers (web and mobile) HOT 1
- [1.x] videoroom: support of string_ids/string_ids_user HOT 2
- [1.x] Increased event handlers SIP logging buffer size one more time or make it configurable HOT 5
- [1.x] build failure on macos HOT 3
- [0.x] Possible memory leaks found with libasan latest HOT 11
- [1.x] SIP plugin "Couldn't connect audio RTP? -- 22 (Invalid argument)" HOT 3
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