I had no idea. Care to share them?
On Fri, Mar 6, 2026, 2:40 AM Andrew Yager <[email protected]> wrote: > Not wanting to hijack my own thread, but have you compiled the g729 codec > support into RTPEngine? There are build options you need to use. > > Andrew > > > On Fri, 6 Mar 2026 at 00:40, Federico Alves <[email protected]> wrote: > >> Also, the RTP engine does not record any calls using the g729 codec. >> Is there something I am missing? >> Philip >> >> On Thu, Mar 5, 2026 at 5:55 AM Andrew Yager <[email protected]> wrote: >> > >> > Hi, >> > >> > I will put money on this being a "you're doing this all the wrong way" >> and you're an idiot type issue. >> > >> > We're running OpenSIPS 3.4.16 with rtpengine 14.0.1.1 and trying to use >> the siprec module with rtp_relay for call recording to a VoIPMonitor SRS. >> > >> > Setup: >> > - rtp_relay_engage("rtpengine") called before siprec_start_recording() >> > - rtp_relay handles the main call's media successfully (offer/answer >> work, RTP flows) >> > - SIPREC INVITE is sent to SRS, SRS responds with 200 OK >> > - BYE sent to SRS immediately (~1ms after ACK) >> > >> > What we see in rtpengine debug logs: >> > >> > The subscribe request sent by rtp_relay uses from-tag: "1" rather than >> the actual call participant's tag: >> > >> > subscribe request: {"call-id": "...", "flags": ["all", "siprec"], >> > "from-tag": "1", "command": "subscribe request"} >> > >> > This succeeds. But the subsequent subscribe answer uses the real callee >> tag: >> > >> > subscribe answer: {"call-id": "...", "from-tag": "ffc01bf7-...", >> > "to-tag": "00e36c57...", "command": "subscribe answer"} >> > >> > rtpengine returns: Failed to process subscription answer >> > >> > It appears the from-tag "1" in the subscribe request is wrong, because >> it should be the actual SIP dialog tag (b2af6068 for caller, ffc01bf7-... >> for callee), and so rtpengine can't correlate the subscribe answer back to >> the subscription because the from-tag changed between request and answer. >> > >> > We're using rtp_relay_engage("rtpengine") followed by >> siprec_start_recording("sip:x.x.x.x:5099") in ROUTE_INVITE after >> create_dialog("B"). Do we need to call siprec_start_recording later, like >> in the reply route on the 200 OK? >> > >> > Is what I'm doing wrong obviously wrong? >> > >> > Thanks, >> > Andrew >> > _______________________________________________ >> > Users mailing list >> > [email protected] >> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> _______________________________________________ >> Users mailing list >> [email protected] >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >
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