[asterisk-users] Webrtc Not acceptable here

[bhavik patel]

Hi Sameer,

Provide me your Asterisk Configuration,may be i can help you.
Also provide me system configuration.


If you need more help then you can post Sipml5 forum
https://groups.google.com/forum/#!forum/doubango.
That way your issue may resolve.



On Wed, Jul 2, 2014 at 8:35 PM, Sameer Rathod <sameer at hostnsoft.com> wrote:

> 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
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>>>
>>
>>
>>
>> --
>> 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
>
>
> --
> _____________________________________________________________________
> -- 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
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