[asterisk-users] Webrtc Not acceptable here
Sameer Rathod
sameer at hostnsoft.com
Thu Jul 3 05:04:35 CDT 2014
I think it is some thing related to strp
Could you please send me your configuration file?
That will be helpful for me.
On Thu, Jul 3, 2014 at 3:25 PM, bhavik patel <bhavikpatel14388 at gmail.com>
wrote:
> Hi Sameer,
>
> I think you should try using public ip rather then local and latest chrome
> browser.
> I have also tried with same configuration and same OS with same asterisk
> version and working fine for me.
>
>
> On Thu, Jul 3, 2014 at 11:59 AM, Sameer Rathod <sameer at hostnsoft.com>
> wrote:
>
>> Hi Bhavik,
>>
>>
>>
>> This is sip.conf
>> [general]
>>
>> context=public
>> allowguest=yes
>> allowoverlap=no
>> realm=192.168.1.151
>> udpbindaddr=0.0.0.0
>> icesupport=yes
>> dtmfmode=rfc2833
>> transport=udp,ws
>> srvlookup=yes
>>
>>
>> [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
>> canreinvite=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
>> ;directmedia=yes ; Asterisk will relay media for this peer
>> transport=udp,ws ; Asterisk will allow this peer to register on UDP or
>> WebSockets
>> canreinvite=yes
>> ;directrtpsetup=yes
>> dtmfmode=rfc2833
>> qualify=yes
>>
>>
>> >> http.conf
>>
>> [general]
>> enabled=yes
>> bindaddr=192.168.1.151
>> bindport=8088
>>
>>
>>
>> >> rtp.conf
>>
>> [general]
>> rtpstart=10000
>> rtpend=20000
>> icesupport=true
>> stunaddr=stun.l.google.com:19302
>>
>>
>> I am using asterisk 12.3 on centos 6.5
>>
>>
>>
>>
>>
>> On Thu, Jul 3, 2014 at 10:31 AM, bhavik patel <bhavikpatel14388 at gmail.com
>> > wrote:
>>
>>> 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
>>>>>>
>>>>>> 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
>>>>
>>>>
>>>> --
>>>> _____________________________________________________________________
>>>> -- 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
>>
>>
>> --
>> _____________________________________________________________________
>> -- 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
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