<div dir="ltr"><div><div>I had also tried with asterisk 11.10.2<br><br></div>no I am getting <br><br>== Using SIP RTP CoS mark 5<br>[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<br>
<br><br><br><br><br></div><div>followed this link <br><a href="http://sipjs.com/guides/server-configuration/asterisk/">http://sipjs.com/guides/server-configuration/asterisk/</a><br></div><div><br></div>following are the configuration I did <br>
<br>[1060] ; This will be WebRTC client<br>type=friend<br>username=1060 ; The Auth user for SIP.js<br>host=dynamic ; Allows any host to register<br>secret=1060 ; The SIP Password for SIP.js<br>encryption=yes ; Tell Asterisk to use encryption for this peer<br>
avpf=yes ; Tell Asterisk to use AVPF for this peer<br>icesupport=yes ; Tell Asterisk to use ICE for this peer<br>context=sameer ; Tell Asterisk which context to use when this peer is dialing<br>directmedia=no ; Asterisk will relay media for this peer<br>
transport=udp,ws ; Asterisk will allow this peer to register on UDP or WebSockets<br><br>[1061] ; This will be the legacy SIP client<br>type=friend<br>username=1061<br>host=dynamic<br>secret=1061<br>context=sameer<br></div>
<div class="gmail_extra"><br><br><div class="gmail_quote">On Thu, Jul 3, 2014 at 3:34 PM, Sameer Rathod <span dir="ltr"><<a href="mailto:sameer@hostnsoft.com" target="_blank">sameer@hostnsoft.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div>I think it is some thing related to strp<br></div><div><br>Could you please send me your configuration file?<br>
</div>That will be helpful for me.<br></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><br><div class="gmail_quote">
On Thu, Jul 3, 2014 at 3:25 PM, bhavik patel <span dir="ltr"><<a href="mailto:bhavikpatel14388@gmail.com" target="_blank">bhavikpatel14388@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr"><div><div>Hi Sameer,<br><br></div>I think you should try using public ip rather then local and latest chrome browser.<br></div>I have also tried with same configuration and same OS with same asterisk version and working fine for me.<br>
</div><div><div><div class="gmail_extra"><br><br><div class="gmail_quote">On Thu, Jul 3, 2014 at 11:59 AM, Sameer Rathod <span dir="ltr"><<a href="mailto:sameer@hostnsoft.com" target="_blank">sameer@hostnsoft.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hi Bhavik,<br><br><br><div><br></div><div>This is sip.conf<br></div><div>[general]<br><br>context=public<br>
allowguest=yes<br>allowoverlap=no<br>realm=192.168.1.151<br>udpbindaddr=0.0.0.0 <br>icesupport=yes<br>
dtmfmode=rfc2833<br>transport=udp,ws<br>srvlookup=yes <br><br><br></div><div><div>[1060] ; This will be WebRTC client<br>
type=friend<br>username=1060 ; The Auth user for SIP.js<br>host=dynamic ; Allows any host to register<br>secret=sameer ; The SIP Password for SIP.js<br>encryption=yes ; Tell Asterisk to use encryption for this peer<br>avpf=yes ; Tell Asterisk to use AVPF for this peer<br>
icesupport=yes ; Tell Asterisk to use ICE for this peer<br>ignorecryptolifetime=yes<br>context=sameer ; Tell Asterisk which context to use when this peer is dialing<br>;directmedia=yes ; Asterisk will relay media for this peer<br>
transport=udp,ws ;Asterisk will allow this peer to register on UDP or WebSockets<br></div>canreinvite=yes<div><br><br>nat=force_rtp,comedia<br>dtmfmode=rfc2833<br>qualify=yes<br><br>
[1061] ; This will be the legacy SIP client<br>type=friend<br>username=1061<br>host=dynamic<br>secret=sameer<br>context=sameer<br>ignorecryptolifetime=yes<br>nat=force_rtp,comedia<br>encryption=yes<br>avpf=yes ; Tell Asterisk to use AVPF for this peer<br>
icesupport=yes ; Tell Asterisk to use ICE for this peer<br></div><div>;directmedia=yes ; Asterisk will relay media for this peer<br>transport=udp,ws ; Asterisk will allow this peer to register on UDP or WebSockets<br>
</div>
canreinvite=yes<br>;directrtpsetup=yes<br>dtmfmode=rfc2833<br>qualify=yes<br><br><br></div><div>>> http.conf<br><br>[general]<br>enabled=yes<br>bindaddr=192.168.1.151<br>bindport=8088<br></div><div><br><br><br></div>
<div>>> rtp.conf<br><br>[general]<br>rtpstart=10000<br>rtpend=20000<br>icesupport=true<br>stunaddr=<a href="http://stun.l.google.com:19302" target="_blank">stun.l.google.com:19302</a><br><br><br></div><div>I am using asterisk 12.3 on centos 6.5<br>
<br><br></div><div><br></div></div><div><div><div class="gmail_extra"><br><br><div class="gmail_quote">On Thu, Jul 3, 2014 at 10:31 AM, bhavik patel <span dir="ltr"><<a href="mailto:bhavikpatel14388@gmail.com" target="_blank">bhavikpatel14388@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div><div><div>Hi Sameer,<br><br></div>Provide me your Asterisk Configuration,may be i can help you.<br>
</div>Also provide me system configuration.<br><br><br></div>If you need more help then you can post Sipml5 forum <a href="https://groups.google.com/forum/#!forum/doubango" target="_blank">https://groups.google.com/forum/#!forum/doubango</a>.<br>
</div>That way your issue may resolve.<br><div><br></div></div><div><div><div class="gmail_extra"><br><br><div class="gmail_quote">On Wed, Jul 2, 2014 at 8:35 PM, Sameer Rathod <span dir="ltr"><<a href="mailto:sameer@hostnsoft.com" target="_blank">sameer@hostnsoft.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div><div>Hi bhavik,<br><br></div>By following the same tutorial <br></div>I am getting this error currently<div>
<br><b>Can't provide secure audio requested in SDP offer<br><br></b></div></div>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 <br>
<div><div><b><br></b></div></div></div><div class="gmail_extra"><div><div><br><br><div class="gmail_quote">On Wed, Jul 2, 2014 at 8:21 PM, bhavik patel <span dir="ltr"><<a href="mailto:bhavikpatel14388@gmail.com" target="_blank">bhavikpatel14388@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div>Hi,<br><br></div>For SIpml5 tried to configure by this way : <a href="https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5" target="_blank">https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5</a><br>
</div>This is working fine for me.<br><br><br></div><div class="gmail_extra"><br><br><div class="gmail_quote"><div><div>On Wed, Jul 2, 2014 at 8:06 PM, Sameer Rathod <span dir="ltr"><<a href="mailto:sameer@hostnsoft.com" target="_blank">sameer@hostnsoft.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><div dir="ltr"><div><div>Hi,<br><br></div>I am getting <br><b>Can't provide secure audio requested in SDP offer</b><br>
<br></div>with sipml5 client hosted on my local system <br clear="all"><div><div><div><br><br><br>[1060] ; This will be WebRTC client<br>
type=friend<br>username=1060 ; The Auth user for SIP.js<br>host=dynamic ; Allows any host to register<br>secret=sameer ; The SIP Password for SIP.js<br>encryption=yes ; Tell Asterisk to use encryption for this peer<br>avpf=yes ; Tell Asterisk to use AVPF for this peer<br>
icesupport=yes ; Tell Asterisk to use ICE for this peer<br>ignorecryptolifetime=yes<br>context=sameer ; Tell Asterisk which context to use when this peer is dialing<br>;directmedia=yes ; Asterisk will relay media for this peer<br>
transport=udp,ws ;Asterisk will allow this peer to register on UDP or WebSockets<br>;disallow=allow<br>;allow=vp8<br>canreinvite=yes<br>;directrtpsetup=yes<br>nat=force_rtp,comedia<br>dtmfmode=rfc2833<br>qualify=yes<br><br>
[1061] ; This will be the legacy SIP client<br>type=friend<br>username=1061<br>host=dynamic<br>secret=sameer<br>context=sameer<br>ignorecryptolifetime=yes<br>nat=force_rtp,comedia<br>encryption=yes<br>avpf=yes ; Tell Asterisk to use AVPF for this peer<br>
icesupport=yes ; Tell Asterisk to use ICE for this peer<br>;context=default ; Tell Asterisk which context to use when this peer is dialing<br>;directmedia=yes ; Asterisk will relay media for this peer<br>transport=udp,ws ; Asterisk will allow this peer to register on UDP or WebSockets<br>
;disallow=allow<br>;allow=vp8<br>canreinvite=yes<br>;directrtpsetup=yes<br>dtmfmode=rfc2833<br>qualify=yes<br><br><br><br><br></div><div>This is my sip.conf <br><br><br></div><div>on the one side I am using zoiper client with 1060 (same pc with ip 192.168.1.191)<br>
</div><div>and for second client I am using sipml5 on chrome <br></div><div><br></div><div>both the client displays a message Not acceptable here <br><br></div><div>I am using asterisk 12.3 <br><br>== WebSocket connection from '<a href="http://192.168.1.191:55561" target="_blank">192.168.1.191:55561</a>' for protocol 'sip' accepted using version '13'<br>
-- Registered SIP '1061' at <a href="http://192.168.1.191:55561" target="_blank">192.168.1.191:55561</a><br> > Saved useragent "IM-client/OMA1.0 sipML5-v1.2014.04.18" for peer 1061<br> == Using SIP RTP CoS mark 5<br>
[Jul 2 19:57:04] WARNING[26672][C-00000071]: chan_sip.c:10648 process_sdp: Can't provide secure audio requested in SDP offer<br><br><br></div><div>If any more information is needed please let me know <br><br></div><div>
My goal is do do peer to peer calling with asterisk+webrtc (i.e. webphone)<span><font color="#888888"><br><br></font></span></div><span><font color="#888888"><div><br><br></div><div><br></div>
<div> <br></div><div><br></div><div><br><br><br>-- <br><div>Regards</div>Sameer Rathod<div>8109413462 <br>
<div><br></div></div>
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<br clear="all"><br>-- <br>Thanks,<br>Bhavik Patel<br>
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<span><font color="#888888">Sameer Rathod<div>
8109413462 <br><div><br></div></div>
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8109413462 <br><div><br></div></div>
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8109413462 <br><div><br></div></div>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br><div>Regards</div>Sameer Rathod<div>8109413462 <br><div><br></div></div>
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