[asterisk-users] packet2packet bridging

[Sameer Rathod]

yes I had configured

icesupport=yes ;

on both the client in sip.con

as well as did the setting of ice in rtp.conf also

here is my sip configuration

[1060] ; This will be WebRTC client
type=friend
username=1060 ; The Auth user for SIP.js
host=dynamic ; Allows any host to register
secret=sameer ; The SIP Password for SIP.js
icesupport=yes ; Tell Asterisk to use ICE for this peer
ignorecryptolifetime=yes
context=sameer ; Tell Asterisk which context to use when this peer is
dialing
;directmedia=yes ; Asterisk will relay media for this peer
transport=udp,ws ;Asterisk will allow this peer to register on UDP or
WebSockets
canreinvite=yes
;directrtpsetup=yes
;nat=force_rtp,comedia
dtmfmode=rfc2833
qualify=yes

[1061] ; This will be the legacy SIP client
type=friend
username=1061
host=dynamic
secret=sameer
context=sameer
ignorecryptolifetime=yes
;nat=force_rtp,comedia
icesupport=yes ; Tell Asterisk to use ICE for this peer
;directmedia=yes ; Asterisk will relay media for this peer
transport=udp,ws ; Asterisk will allow this peer to register on UDP or
WebSockets
canreinvite=yes
dtmfmode=rfc2833
qualify=yes




On Wed, Jul 2, 2014 at 8:00 PM, Joshua Colp <jcolp at digium.com> wrote:

> 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
>
> --
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-- 
Regards
Sameer Rathod
8109413462
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