[asterisk-users] packet2packet bridging

[Joshua Colp]

Sameer Rathod wrote:
> = Using SIP RTP CoS mark 5
>      -- Executing [1061 at sameer:1] Dial("SIP/1060-00000088", "SIP/1061")
> in new stack
>    == Using SIP RTP CoS mark 5
>      -- Called SIP/1061
>      -- SIP/1061-00000089 is ringing
>  > 0x7f67f90b43c0 -- Probation passed - setting RTP source address to
> 192.168.1.176:8000 <http://192.168.1.176:8000>
>      -- SIP/1061-00000089 answered SIP/1060-00000088
>      -- Channel SIP/1061-00000089 joined 'simple_bridge' basic-bridge
> <1937a88f-5b6f-4a5a-ae3c-adc2d68c8bb4>
>      -- Channel SIP/1060-00000088 joined 'simple_bridge' basic-bridge
> <1937a88f-5b6f-4a5a-ae3c-adc2d68c8bb4>
>  > Bridge 1937a88f-5b6f-4a5a-ae3c-adc2d68c8bb4: switching from
> simple_bridge technology to native_rtp
>  > 0x7f67f90b43c0 -- Probation passed - setting RTP source address to
> 192.168.1.176:8000 <http://192.168.1.176:8000>
>  > 0x7f6780047090 -- Probation passed - setting RTP source address to
> 192.168.1.191:8000 <http://192.168.1.191:8000>
>    == WebSocket connection from '192.168.1.191:54390
> <http://192.168.1.191:54390>' closed

Are either side using encryption or ICE?

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org