04 July 2013

Recently I've been trying to get a web phone up and running, my only real requirement was to use Asterisk. So I decided to go with the following technology stack, JsSIP, Chrome and Asterisk.


I've only tried to use chrome so far, though I've read that Firefox is currently WebRTC capable as well.

Chrome on OSX seems to work fine (version 27.0.1453.116).

Chrome on Ubuntu had problems until I updated to a Beta version (version 28.0.1500.52 beta)


Asterisk had quite a few requirements before I could get it to work with WebRTC, see this page for details.


You will likely need to rebuild Asterisk as WebRTC requires a SRTP libraries, which aren't included by default.

You will need the following libraries on the machine you use to recompile Asterisk:

  • libssl-dev
  • libsrtp0

Once you have these libraries installed, you will also need to enable two Asterisk resources:

  • res_srtp.so
  • res_http_websocket.so

Now you can recompile.

The resulting build might also need a few configuration changes:

  |-- http.conf
  |-- modules.conf
  |-- sip.conf
  • http.conf Ensure the following options are set

  • modules.conf you must load res_http_websocket.so before chan_sip.so

  • sip.conf Any users that you want to allow WebRTC for you need:


    Adding encryption=yes to any non WebRTC phones might make them break, so be careful.

Remember to restart Asterisk once you're done.


JsSIP JsSIP was quite easy to use, however it wasn't without its set of problems. If you want to do a quick test yourself, check out the JsSIP Tryit page.

My main problem was that their script didn't seem to connect with asterisk properly, though I've already forgotten the reason (will update if I do). To solve this I updated to the dev version of JsSIP, which I download from the JsSIP Tryit page.

Afterwards I would also have jssip error out when I tried to type in an invalid target, I patched it quickly removing the potentially erroneous code. I didn't know enough about their side of the problem, so I have no way of knowing if this is a correct fix.

I've included the patch below, use it at your own risk.

Index: /static/js/jssip-devel.js
--- /static/js/jssip-devel.js
+++ /static/js/jssip-devel.js
@@ -3372,5 +3372,10 @@
         console.log(LOG_PREFIX +'ICE candidate received: '+ e.candidate.candidate);
       } else {
-        self.onIceCompleted();
+       // PATCH: (saevon) Fixes bug with the code crashing at this step.
+       // since: neither createOffer nor createAnswer get called if you
+       // had an "Invalid Target"
+       // Thus the method doesn't get added yet
+       if (self.onIceCompleted) {
+           self.onIceCompleted();
+       }


WebRTC is clearly still a work in progress, and I hope it gets polished up nicely for when I next wish to use it.


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