SIPML5 Asterisk Example

[arpit modi]

Hi All,

I am trying to explore an example for SIPml5 on my system.

I have asterisk 12.1.1 installed and downloaded source of SIPml5 from http://code.google.com/p/sipml5/source/checkout
I copied sample code into web root directory and example loaded successfully.

I also configured SIP users correctly in sip.conf but when i try to register SIP extension using http://127.0.0.1/sipml5/call.htm?svn=220 on my machine, extension does not get registered.

Here is my sip.conf entry:

[8888]
username=8888
secret=8888
type=friend
qualify=yes
nat=yes,force_rport
host=dynamic
context=testcall
canreinvite=no
transport=ws
avpf=yes
encryption=yes


Can anyone please let me know what am i doing wrong?


Thanks,
Arpit

[arpit modi]

Hi,

Can anyone please suggest something on this?

Thanks

[navaismo]

You need to configure the websocket via htttp.conf in asterisk, add the icesupport to your peer, if you are using PJSIP leave the transport blank.

And finally provide logs from Chrome and asterisk to see whats your issue.

[arpit modi]

Hi,

Thanks for your response.

Already i have enabled http websocket in http.conf , here is my asterisk CLI output

CLI> http show status
HTTP Server Status:
Prefix:
Server Enabled and Bound to 127.0.0.1:8088

Enabled URI's:
/httpstatus => Asterisk HTTP General Status
/phoneprov/... => Asterisk HTTP Phone Provisioning Tool
/static/... => Asterisk HTTP Static Delivery
/ari/... => Asterisk RESTful API
/ws => Asterisk HTTP WebSocket

Enabled Redirects:
  None.

Is there any other config needs to be done in http.conf ?

Also i added icesupport=yes in my sip.conf entry but still no luck :(

I am just running standard install of Asterisk 12.1.1

Is there anything still missing?

Thanks,
Arpit Modi

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[navaismo]

You need to provide logs from both asterisk and chrome and possibly your listen port is wrong in the http.conf try with 0.0.0.0

[aishwarya belide]

hi, i am also working on webrtc and asterisk i got the similar problem. if you got the solution please post a reply aishwar...@gmail.com

Thank you

On Saturday, 12 April 2014 15:34:30 UTC+5:30, arpit modi wrote:

[Tom Chandler]

I use transport = udp,ws and icesupport=yes and it works for me.  All of the other values look ok.

Tom C

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[arpit modi]

Hi,

That's actually strange case.

If i setup on a server with public IP and use that as domain and realm then at least SIP registration works fine but having issue with calling still.  So i am checking more.

I never got it worked on a local system which is having asterisk installed. I tried both 127.0.0.1 / localhost and local IP (192.168.0.5)

Thanks,
Arpit Modi

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[Mamadou DIOP]

even if SIPML5 is remotely hosted (sipml5.org) the webrtc stack is running on your local machine. No need to host your own sipml5 code to connect to a private/public asterisk.

[arpit modi]

Yeah i did the same. SipML5 demo is in my local machine which i am accessing using 127.0.0.1 and registering SIP user of my public IP server.

Thanks,
Arpit Modi

[Mamadou DIOP]

Not what I meant. I asked to keep using sipml5.org.

[arpit modi]

Hi,

Yes using demo provided on sipml5.org, its same.

Actually i am trying to modify the demo code and integrate into a php page, where i can dial numbers. But still not succeeded even in default demo code.

Thanks,
Arpit Modi

[Mamadou DIOP]

My response was just about using local sipml5 or not. Not about your issue. Without useful technical information and logs few chances to get help/response about your issue.

[arpit modi]

Yes, I totally agree.
As said by navaismo above, I can get asterisk logs. But i am not sure what exactly he mean by chrome logs?

Please help me with the technical information and logs needed for this.

Thanks,
Arpit Modi

[Yusuf Siddiqui]

You need to know Chrome way of page debugging webrtc then you may get logs. Read webrtc debugging,it will help you.

Regards

Mohd Yusuf Siddiqui
(Sr.Program Manager)
email: yusuf.s...@fiyutech.com

www.fiyutech.com

Mob. +91.995.899.521.8

Off:+91.120.437.209.3

U.S. +120.975.347.57

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[arpit modi]

Thanks, I will checkout and get the logs.

Thanks,
Arpit Modi

[navaismo]

Check this--> http://forums.digium.com/viewtopic.php?f=1&t=90167&sid=2c25bc30a1b2b8a9c15a10ae706c5ffe