[asterisk-users] Webrtc Not acceptable here

[bhavik patel]

Hi,

For SIpml5 tried to configure by this way :
https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5
This is working fine for me.




On Wed, Jul 2, 2014 at 8:06 PM, Sameer Rathod <sameer at hostnsoft.com> wrote:

> 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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-- 
Thanks,
Bhavik Patel
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