[asterisk-users] Webrtc Not acceptable here
Sameer Rathod
sameer at hostnsoft.com
Thu Jul 3 01:29:59 CDT 2014
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
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