[Freeswitch-users] problem with eavesdrop() sending null RTP for 15-30 seconds
Nathan Neulinger
nneul at mst.edu
Thu Nov 20 23:26:56 MSK 2014
Seeing an interesting issue with sending an eavesdrop where the outbound call gets established, but no useful audio is
transmitted until around 15-30 seconds into the call.
Prior to that point, FS is transmitting all identical RTP frames with the payload containing:
24 1c 61 7a 7f 55 55 55 50 06 83 ff ff ff ff ff ff ff ff ff ff ff ff
aa aa aa aa aa aa aa aa aa aa aa aa aa aa aa aa aa aa aa aa aa aa af
ff ..... ff
This is minimal version of command that was being issued:
originate sofia/internal/6679 at voice-dev.mst.edu &eavesdrop(ca887b3b-013f-4972-aa7c-37a7b132bf52)
Reproduced with head of master as of a few minutes ago. Restarting FS and trying again, the FIRST attempted eavesdrop
seems to not have the long delay, but the second one does. The 'dummy' payload appears to have the same content in
subsequent requests.
Interestingly, if I let the outbound call ring longer before I pick up, the delay appears to be reduced by the period of
time I let the phone ring for. if I pick up immediately, 25-30 second audio delay approx, if I pick up after 25 seconds
of ringing, I get only a couple second delay.
Any idea where to start looking for where this buffering might be getting introduced?
For the "only eavesdrop" case it's not as big of an issue, but when you switch over to barge, can't really have
multi-second audio delay.
-- Nathan
------------------------------------------------------------
Nathan Neulinger nneul at mst.edu
Missouri S&T Information Technology (573) 612-1412
System Administrator - Architect
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