FreePBX settings for HTML5 SIP client

I wish to create SIP client for my WEB APPLICATION. To implement SIP client for the WEB page what are the settings needed in FreePBX server side ?

Is FreePBX Server has any default SIP Client for WEB ?

I just have the FreePBX Server only, To achieve SIP Client on WEB Page need any additional installation ?

Is there any open source client available for my need ?

If you can afford to wait until “Progressive Web Apps” are available in a widespread way, that sounds like the way to go.

Thanks for your response @cynjut

I need to implement SIP Client on my web application only, that is my primary need.

With the help of open sources like SIPML5 or JSSIP. I wish to implement it on my web application.

For this need, what are the possibilities available in FreePBX server ?

Is there any settings is need additional ? For example, i had used UDP port 5060 to connect with my android client, call is working perfectly in Android. To achieve this with the Web application, instead of UDP is there any need of other protocol ? instead of 5060 port is any other port is needed for the particular extension.

I am very new to this platform, Guide me to do.
Thanks in advance.

If your web app is actually a SIP client, the configuration of the interface to the PBX will be exactly the same as in your Android client.

The primary difference is going to be that the people that wrote the Android app understand how the SIP protocol works - it’s not like a web page or any other service. There are lots of parts and pieces that all have to work correctly. You will need access to the SIP port on your server (UDP ports 5060 or 5160, typically), plus you will need to set up RTP access to the server.

I don’t understand the context of this question. What is it that you are really asking? The possibilities to solve this in FreePBX don’t exist, because FreePBX is a web-based management suite for Asterisk. It’s like asking for a Pina Colada at your son’s daycare.

Thanks for your reply Dave Burgess

My RTP port range 10000 to 20000.
For all my extension i had used Protocols - UDP as primary.
I had created Extensions only under Chan_sip only.

I had used the ctxSIP client

When I try to register with my client, on FreePBX server PC, Asterisk shows the following error logs

handle_tcptls_connection: Problem setting up ssl connection: error: 00000001: lib(0);func(0):reason(1), Internal SSL error

For my web client I had used the below information,

User: 151
password : [email protected]

For this need what are the possibilities or settings are needed for the FreePBX server ?
I am using self-Signed SSL certificate (Default one)

The WSS stuff might be jacking you up.

There are some issues with WSS that we’ve discussed here in the past. Perhaps a quick search through the group might help you.

I’m suspicious of your SSL configuration. For example, I don’t see anywhere that you’ve set up the TLS stuff correctly in the server. We’ve seen problems with this as well in the past few months.

It’s entirely possible that your client software is working correctly. Double check the SIP TLS stuff - it’s not intuitively simple to set up and there are many moving parts that need to be reviewed. Note that I am no expert when it comes to the Security and HTTPS parts of the system. You might need some help from one of the heavyweights, and the simplest way to do that is to look back through the forums looking for WSS problems and solutions.

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