[asterisk-users] Webrtc Not acceptable here

[Sameer Rathod]

Hi,

I am getting
*Can't provide secure audio requested in SDP offer*

with sipml5 client hosted on my local system



[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=yes




This is my sip.conf


on 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 chrome

both the client displays a message Not acceptable here

I 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 offer


If any more information is needed please let me know

My goal is do do peer to peer calling with asterisk+webrtc (i.e. webphone)









-- 
Regards
Sameer Rathod
8109413462
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