Displaying 2 results from an estimated 2 matches for "force_rtp".
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force_ro
2014 Jul 02
1
Webrtc Not acceptable here
...tolifetime=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
;disallow=allow
;allow=vp8
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
encryption=yes
avpf=yes ; Tell Asterisk to use AVPF for this peer
icesupport=yes ; Tell Asterisk to us...
2014 Jul 02
1
packet2packet bridging
Hi,
I am new to asterisk I want to configure my asterisk server such that it
only establishes the call
rest the audio must bypass the server and transmitted directly to the peer
In my config file I did changes which are below
canreinvite=yes
nat=force_rtp
dirtectmedia=yes
directsetup=yes
I am using asterisk version 12.3
--
Regards
Sameer Rathod
8109413462
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