[asterisk-users] Webrtc Not acceptable here
bhavik patel
bhavikpatel14388 at gmail.com
Thu Jul 3 04:55:30 CDT 2014
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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140703/747465fe/attachment.html>
More information about the asterisk-users
mailing list