Browser to browser VoIP calling using SIP? [closed]

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Closed 8 years ago.

开发者_运维技巧 Improve this question

I am trying to call from one browser to another browser.

What I have done:

1. I set up my asterisk server just like getonsip.com
2. I wrote a client using pjsua python binding which acts as a server

Here is the situation:

1. I can call from my browser ( My browser is on the same machine on which pjsua client is running) to my sip client on my android phone.

2. I am not able to do same from the browser from another machine.

3. What should I do for making a browser to browser call happen?

4. Does streaming and flash required for doing this?

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You won't be able to do browser-to-browser calls with SIP at least not without a lot of plugin magic. The first problem is that by default SIP uses UDP port 5060, the browser sandbox won't let you communicate on that port so you'd need a plugin (Flash, Silverlight, Java Applet etc.). Once you've overcome the SIP side, and dealt with the NAT headaches along the way, you'd then need to work out how to get the browser to process the RTP media stream. That would mean more plugins.

There is actually an upcoming standard that deals with exactly what you are after, see RTC-WEB. While you're waiting for that you could try and get by with phono which is a javascript/flash component from Tropo that supports browser based calling but with the caveat being the call will traverse Tropo's SIP servers.

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It's possible now, since WebRTC finally became standard and there are already implementations in both browsers and SIP server

• Asterisk 11 is officially support WebRTC
• Google Chrome has built-in WebRTC support since version 23

See this tutorial for more details http://www.html5rocks.com/en/tutorials/webrtc/basics/