[asterisk-dev] asterisk's bridge

[serva]

"Is RTP brige working in Asterisk 1.4.x?" I think must define P2P_INTENSE, and do other configure.

"I know SIP <--> SIP always works with RE-INVITE is on (i.e.RTP streams travel 
from endpoint to endpoint) in Asterisk 1.2.x.  "  Yes, if you want to let RTP travel from endpoint to endpoin always need RE-INVITE  in Asterisk 1.2.x.

"SIP <--> H323 or SCCP/Skinny was not working (i.e. RTP streams travel through 
Asterisk) in Asterisk 1.2.x. How about 1.4.x? Is there any improvement? " 

First, It can work  in Asterisk 1.2.x.
Second, I think the problem is muti-threads, so asterisk need a thread deal with all the channel in one bridge. If we need mixmonitor or spy and so on, we can let bridge return to generic bridge,so the single-thread will force to become muti-thread, if we stop mixmonitor, this call can return to sigle-thread bridge.   

 

------------------				 
serva
2007-11-23

-------------------------------------------------------------
From：K C
Date：2007-11-23 01:08:35
Subject：Re: [asterisk-dev] asterisk's bridge

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