This is working fine for me.Hi,For SIpml5 tried to configure by this way : https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5
On Wed, Jul 2, 2014 at 8:06 PM, Sameer Rathod <sameer@xxxxxxxxxxxxx> wrote:
--with sipml5 client hosted on my local systemHi,I am getting
Can't provide secure audio requested in SDP offer
[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
encryption=yes ; Tell Asterisk to use encryption for this peer
avpf=yes ; Tell Asterisk to use AVPF for this peer
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
;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 use ICE for this peer
;context=default ; 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
dtmfmode=rfc2833
qualify=yesThis is my sip.confon the one side I am using zoiper client with 1060 (same pc with ip 192.168.1.191)
and for second client I am using sipml5 on chromeboth the client displays a message Not acceptable hereI am using asterisk 12.3
== WebSocket connection from '192.168.1.191:55561' for protocol 'sip' accepted using version '13'
-- Registered SIP '1061' at 192.168.1.191:55561
> Saved useragent "IM-client/OMA1.0 sipML5-v1.2014.04.18" for peer 1061
== Using SIP RTP CoS mark 5
[Jul 2 19:57:04] WARNING[26672][C-00000071]: chan_sip.c:10648 process_sdp: Can't provide secure audio requested in SDP offerIf any more information is needed please let me knowMy goal is do do peer to peer calling with asterisk+webrtc (i.e. webphone)
--RegardsSameer Rathod8109413462
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Thanks,
Bhavik Patel
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