[Asterisk-Users] How to make * send RTCP reports
HT
ht-lists at softhome.net
Fri Jul 4 10:38:10 MST 2003
Hi,
I am plying with * for 10 days now. I am testing with a couple of vocaltec
h.323 gateways (FXO and PRI) cisco ata-186 (configured for SIP) and MSN
messenger (SIP). They all seem to interoperate. However I have a problem
when * is sending calls to the vocaltec gateways. Vocaltec gateways are
monitoring the RTCP reports send from the remote gateway (in this case *)
and if they don't get a report for 60 seconds they will disconnect the call
(assuming internet disconnection). Because of this all my calls have
duration of one minute.
I can see on the console that * is detecting incoming RTCP reports so there
should be some RTCP functionality in it (although I have seen a message from
February saying the opposite). My question is if/how can I make * send RTCP
report to the vocaltec gateways. I think any RTCP packet will do the trick
as long as the vocaltec gateway gets it on a regular basis (I don't care if
the information in it is correct).
H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030704/3c2d4682/attachment.htm
More information about the asterisk-users
mailing list