[asterisk-users] Webrtc Not acceptable here
Sameer Rathod
sameer at hostnsoft.com
Thu Jul 3 05:18:28 CDT 2014
I had also tried with asterisk 11.10.2
no I am getting
== Using SIP RTP CoS mark 5
[Jul 3 15:45:10] WARNING[29686][C-00000001]: chan_sip.c:10509 process_sdp:
Rejecting secure audio stream without encryption details: audio 9191
UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
followed this link
http://sipjs.com/guides/server-configuration/asterisk/
following are the configuration I did
[1060] ; This will be WebRTC client
type=friend
username=1060 ; The Auth user for SIP.js
host=dynamic ; Allows any host to register
secret=1060 ; 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
context=sameer ; Tell Asterisk which context to use when this peer is
dialing
directmedia=no ; Asterisk will relay media for this peer
transport=udp,ws ; Asterisk will allow this peer to register on UDP or
WebSockets
[1061] ; This will be the legacy SIP client
type=friend
username=1061
host=dynamic
secret=1061
context=sameer
On Thu, Jul 3, 2014 at 3:34 PM, Sameer Rathod <sameer at hostnsoft.com> wrote:
> 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
>
>
--
Regards
Sameer Rathod
8109413462
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