[asterisk-users] Webrtc Not acceptable here

[Sameer Rathod]

Hi bhavik,

By following the same tutorial
I am getting this error currently


*Can't provide secure audio requested in SDP offer*
I think it is related to the srtp issue of asterisk Please help me in this
I am struggling with this form a long time



On Wed, Jul 2, 2014 at 8:21 PM, bhavik patel <bhavikpatel14388 at gmail.com>
wrote:

> 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
>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>                http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Thanks,
> Bhavik Patel
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Regards
Sameer Rathod
8109413462
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140702/3d51bdf6/attachment.html>