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[sameer at hostnsoft.com]

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 Rathod8109413462 

[bhavikpatel14388 at gm...]

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@hostnsoft.com (sameer@hostnsoft.com)> wrote:

[sameer at hostnsoft.com]

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@gmail.com (bhavikpatel14388@gmail.com)> wrote:

[bhavikpatel14388 at gm...]

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 [url=https://groups.google.com/forum/#!forum/doubango]https://groups.google.com/forum/#!forum/doubango[/url].

That way your issue may resolve.





On Wed, Jul 2, 2014 at 8:35 PM, Sameer Rathod <sameer@hostnsoft.com (sameer@hostnsoft.com)> wrote:

[sameer at hostnsoft.com]

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

[bhavikpatel14388 at gm...]

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@hostnsoft.com (sameer@hostnsoft.com)> wrote:

[sameer at hostnsoft.com]

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@gmail.com (bhavikpatel14388@gmail.com)> wrote:

[sameer at hostnsoft.com]

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@hostnsoft.com (sameer@hostnsoft.com)> wrote:

[sameer at hostnsoft.com]

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.onSetRemoteDescriptionErrortmedia_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.onCreateSdpErrortmedia_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.onIceCandidatetmedia_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@hostnsoft.com (sameer@hostnsoft.com)> wrote: