[asterisk-users] Webrtc Not acceptable here
Sameer Rathod
sameer at hostnsoft.com
Thu Jul 3 08:50:14 CDT 2014
This one is not fully related but
with asteerisk 11.9.0 and webrtc sipml5 client
I am getting this on client side
1. Failed to set remote offer sdp: Called with SDP without DTLS
fingerprint. tsk_utils.js?svn=224:128
1. tsk_utils_log_errortsk_utils.js?svn=224:128
2. tmedia_session_jsep01.onSetRemoteDescriptionError
tmedia_session_jsep.js?svn=224:644
3. (anonymous function)tmedia_session_jsep.js?svn=224:789
1. CreateAnswer can't be called before SetRemoteDescription.
tsk_utils.js?svn=224:128
1. tsk_utils_log_errortsk_utils.js?svn=224:128
2. tmedia_session_jsep01.onCreateSdpError
tmedia_session_jsep.js?svn=224:605
3. (anonymous function)tmedia_session_jsep.js?svn=224:562
1. This/PeerConnection is null: unexpected tsk_utils.js?svn=224:128
1. tsk_utils_log_errortsk_utils.js?svn=224:128
2. tmedia_session_jsep01.onIceCandidate
tmedia_session_jsep.js?svn=224:677
3. o_pc.onicecandidate
On the other side I had used blink as a second client
and enabled DTLS-SRTP setting
any idea why this happens??
On Thu, Jul 3, 2014 at 3:48 PM, Sameer Rathod <sameer at hostnsoft.com> wrote:
> 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
>
>
--
Regards
Sameer Rathod
8109413462
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