Re: Webrtc Not acceptable here

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

 



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@xxxxxxxxx> 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@xxxxxxxxxxxxx> 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

[Index of Archives]     [Asterisk Announcements]     [Asterisk App Development]     [PJ SIP]     [Gnu Gatekeeper]     [IETF Sipping]     [Info Cyrus]     [ALSA User]     [Fedora Linux Users]     [Linux SCTP]     [DCCP]     [Gimp]     [Yosemite News]     [Deep Creek Hot Springs]     [Yosemite Campsites]     [ISDN Cause Codes]     [Asterisk Books]