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sipml5 - Asterisk.wiki

We strongly recommend using the RTCWeb Breaker (requires http://webrtc2sip.org'>webrtc2sip between the browser and Asterisk) on the http://sipml5.org/expert.htm'>expert window instead of patching Asterisk. Using http://webrtc2sip.org'>webrtc2sip with the Media Coder enabled also allows video transcoding (e.g. VP8 <-> H.264).
For more info on how to use webrtc2sip and Asterisk: http://linux.autostatic.com/asterisk-and-sipml5-interoperability'>http://linux.autostatic.com/asterisk-and-sipml5-interoperability

This is a short guide explaining how to configure any HTML5 SIP client to work with Asterisk. This example talk about sipml5 client but any client using WebRTC should work. We also use xlite5 but off course any SIP client could be used. I assume that you already know how Asterisk works.

http://sipml5.googlecode.com/svn/trunk/images/architecture_asterisk.png
sipML5 solution using Asterisk

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Building source code

The current Asterisk version (revision 379070) allows to register and make calls from Chrome but no audio or video will flow. The problem comes from ICE implementation in Chrome which is not fully compliant with RFC 5245. To fix this problem you'll need to apply the provided patch and rebuild Asterisk as explained in this section. This patch also adds experimental support for VP8 video codec.

  1. checkout Asterisk source code revision 379070 svn checkout -r 379070 http://svn.digium.com/svn/asterisk/trunk asterisk
  2. copy the patch into asterisk folder cd asterisk wget http://sipml5.googlecode.com/svn/trunk/asterisk/asterisk_379070.patch
  3. apply the patch patch -p0 -i ./asterisk_379070.patch
  4. configure for build export PREFIX=/opt/asterisk ./configure --with-crypto --with-ssl --with-srtp --prefix=$PREFIX
  5. build and install make make install make samples

Configure sip.conf

Open $PREFIX/etc/asterisk/sip.conf and change the default values as follow udpbindaddr=0.0.0.0:5060 realm=doubango.org transport=udp,ws,wss

Configure http.conf

Open $PREFIX/etc/asterisk/http.conf and change the default values as follow enabled=yes bindaddr=0.0.0.0 bindport=8088

Configure rtp.conf

Open $PREFIX/etc/asterisk/rtp.conf and change the default values as follow stunaddr=stun.l.google.com:19302

Create two users for testing

Open $PREFIX/etc/asterisk/users.conf and add two users (1060 for chrome and 1061 for xlite5) ``` [1060] type=peer username=1060 host=dynamic secret=1060 context=default hasiax = no hassip = yes encryption = yes avpf = yes icesupport = yes videosupport=no directmedia=no

[1061] type=peer username=1061 host=dynamic secret=1061 context=default hasiax = no hassip = yes ; enable ice if supported by your ua (e.g. xlite) icesupport = yes ```

Setting extensions

Open $PREFIX/etc/asterisk/extensions.conf and add two extensions to the default section exten => 100,1,Dial(SIP/1060) exten => 101,1,Dial(SIP/1061)

Start Asterisk

To start Asterisk $PREFIX/sbin/safe_asterisk

Configure sipml5

  • Open http://sipml5.org/call.htm on your browser
  • Click on Expert mode? and set the fields as follow http://sipml5.googlecode.com/svn/trunk/asterisk/screenhot_chrome_expert.png
    Expert mode fields settings
    The websocket connection url is ws://192.168.0.12:8088/ws (Do not forget the /ws at the end). For secure connection use wss:// instead of ws://
    192.168.0.12 = Asterisk Server IP address. You have to change this value with yours.
    8088 = Websocket listening port (defined in $PREFIX/etc/asterisk/http.conf)

  • Go back to the home screen and set the fields as follow http://sipml5.googlecode.com/svn/trunk/asterisk/screenshot_chrome_credentials.png
    Credentials fields settings
    The password value is 1060 as defined in $PREFIX/etc/asterisk/users.conf

  • Press LogIn to connect to the server

Configure Xlite5

  • Download xlite5 from http://www.counterpath.com/x-lite.html and Install it
  • Add new Account (Preferences -> Accounts -> Add) and configure the credentials as follow http://sipml5.googlecode.com/svn/trunk/asterisk/screenshot_xlite5_credentials.png
    xlite5 account configuration
    192.168.0.12 = Asterisk Server IP address. You have to change this value with yours.
    5060 is the default port defined in $PREFIX/etc/asterisk/sip.conf
  • Enable Firewall Traversal (Topology tab) as follow http://sipml5.googlecode.com/svn/trunk/asterisk/screenshot_xlite5_natt.png
    Firewall traversal

Testing with chrome

From sipml5 home page, enter 1061 in the call control box and press call.

Testing with Firefox, IE9+, Safari and Opera

To test with these browsers you will need to install webrtc4all version 1.11.745 or later.
IMPORTANT: The websocket transport is disabled on these browsers (because not supported or using old version) and you have to set a SIP outbound proxy using the expert view. Only UDP is supported as transport protocol. The value of the SIP outbound proxy should be something like udp://192.168.0.12:5060.

Experimental support for VP8 video codec

Starting r106 there is experimental support for VP8 video codec. To be used at your own risk.

Updates

  1. r90: Fix one-way audio when the called party is chrome
  2. r91: Fix ICE authentication issues
  3. r98: Add support for Firefox, IE9+, Safari and Opera (thanks to webrtc4all). Disable STUN keepAlive to avoid overrun.
  4. r104: Fix issue 42. Use Asterisk 373330. Re-enable STUN keepAlive.
  5. r106: Adds support for VP8 video codec in Asterisk.
  6. r169: Update patch (tested with chrome stable 24.0.1312.52 m and canary 26.0.1384.2)

Technical help

Please check our issue tracker or developer group if you have any problem.

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bossiel