WebRTC with Asterisk 14 on Centos 6.6


Need support to configure WebRTC with Asterisk 14.

When making job offers, you need to indicate how much you are prepared to pay. I’m not currently in that market.

If you want free support, you need to provide details of where you are getting stuck, and what you have tried.

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There is a tutorial on the wiki[1] which provides examples of how to configure things.

[1] https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5

We want free support.

We have configured the WebRTC with the help of below link.

In that I have done up to exensions.conf . After that I am not getting how to use that with call.htm

We have followed that link also but unable to login. Can you please help us to configure WebRTC with Asterisk.

We are using the server locally and given the local subnet IP in realm.

If you provide more detail about what exactly isn’t working then someone here may provide suggestions.

I have configured the WebRTC with the help of below link and link provided by you previously.


I have configured till extensions.conf and then there is a htm page like call.htm . Can you please help me what is that htm page and how to write that htm page.

No, that’s equivalent to asking someone to write a SIP client for you for free. It’s unlikely anyone would ever do that. There are options out there can be used for testing or as a base, such as SIPml5 or JSSIP. If you want to turn those into a custom client you would need to find a web developer.

I want to do testing to call using SIPml5 but I am not able to login through below link.


That’s not enough information to help. Does Asterisk reject the attempt? Does it even see the connection attempt? What does the browser say? Does it register but you can’t call?

Getting below error while login .

[sip;6001@] is not a valid Public identity

That would be because you are using “;” instead of “:” in the SIP URI. It should be "sip:6001@"

Now I was login through but when I am calling it is showing that Media stream permission denied error.

That would be a browser issue, for example you not accepting the request to allow access to media.

Getting below error when calling between sip agents.

[May 4 16:54:08] NOTICE[10410][C-0000000b]: chan_sip.c:26328 handle_request_invite: Failed to authenticate device sip:6001@X.X.X.X;tag=60429874600

The device at 6001 sent the wrong password.

It’s pretty clear. It failed to authenticate. You have to determine what username it is using and confirm that is correct. Then afterwards confirm the password is correct. This is all basic troubleshooting stuff. If you plan on using WebRTC at all you are going to need to gain troubleshooting skills yourself. It’s not something that generally “just works”.

No it was logged in successfully. But when calling another sip agent we are getting that error.

SIP devices register. They don’t login.

In any case, this error refers to handle request invite, which means it is the incoming leg of the call.