Update BUILDING, CHANGELOG and README files

jmillan · jmillan · commit 5c754b4e288e · 2013-03-18T22:26:29.000+01:00
diff --git a/BUILDING.md b/BUILDING.md
@@ -15,7 +15,7 @@ You just need to have [Node.js](http://nodejs.org/) and [Git](http://git-scm.com
 
 ### PhantomJS
 
-(optional, just for running test units)
+(optional, just for running unit tests)
 
 * [Install PhantomJS](http://phantomjs.org/download.html)
 * In modern Debian/Ubuntu systems PhantomJS can be installed via `apt-get install phantomjs`
@@ -51,9 +51,9 @@ $ grunt
 The built version of JsSIP will be available in the `dist/` subdirectory in both flavors: normal (uncompressed)  and minified, both linted with [JSLint](http://jslint.com/). There will be also a file named `dist/jssip-devel.js` which is an exact copy of the uncompressed file.
 
 
-## Faster development
+## Development version
 
-Run `grunt devel` for just generating the `dist/jssip-devel.js` file.
+Run `grunt devel` for just generating the `dist/jssip-devel.js` file. An uncompressed JsSIP source file named `jssip-devel.js` will be created in `dist` directory.
 
 
 ## Test units
@@ -73,4 +73,4 @@ If you modify `src/Grammar/src/Grammar.pegjs` then you need to recompile JsSIP g
 ```
 $ grunt grammar
 ```
-And then build JsSIP again by running `grunt`.
+And then build JsSIP again as explained above.
diff --git a/CHANGELOG.md b/CHANGELOG.md
@@ -2,6 +2,84 @@ CHANGELOG
 =========
 
 
+Version 0.3.0 (released in 2013-03-18)
+-------------------------------
+
+* [(fea1326)](https://github.com/versatica/JsSIP/commit/fea1326) Don't validate configuration.password against SIP URI password BNF grammar (fix #74).
+* [(3f84b30)](https://github.com/versatica/JsSIP/commit/3f84b30) Make RTCSession local_identity and remote_identity NameAddrHeader instances
+* [(622f46a)](https://github.com/versatica/JsSIP/commit/622f46a) remove 'views' argument from UA.call()
+* [(940fb34)](https://github.com/versatica/JsSIP/commit/940fb34) Refactored Session
+* [(71572f7)](https://github.com/versatica/JsSIP/commit/71572f7) Rename causes.IN_DIALOG_408_OR_481 to causes.DIALOG_ERROR and add causes.RTP_TIMEOUT.
+* [(c79037e)](https://github.com/versatica/JsSIP/commit/c79037e) Added 'registrar_server' UA configuration parameter.
+* [(2584140)](https://github.com/versatica/JsSIP/commit/2584140) Don't allow SIP URI without username in configuration.uri.
+* [(87357de)](https://github.com/versatica/JsSIP/commit/87357de) Digest authentication refactorized.
+* [(6867f51)](https://github.com/versatica/JsSIP/commit/6867f51) Add 'cseq' and 'call_id' attributes to OutgoingRequest.
+* [(cc97fee)](https://github.com/versatica/JsSIP/commit/cc97fee) Fix. Delete session from UA sessions collection when closing
+* [(947b3f5)](https://github.com/versatica/JsSIP/commit/947b3f5) Remove RTCPeerConnection.onopen event handler
+* [(6029e45)](https://github.com/versatica/JsSIP/commit/6029e45) Enclose every JsSIP component with an inmediate function
+* [(7f523cc)](https://github.com/versatica/JsSIP/commit/7f523cc) JsSIP.Utils.MD5() renamed to JsSIP.Utils.calculateMD5() (a more proper name for a function).
+* [(1b1ab73)](https://github.com/versatica/JsSIP/commit/1b1ab73) Fix. Reply '200' to a CANCEL 'before' replying 487 to the INVITE
+* [(88fa9b6)](https://github.com/versatica/JsSIP/commit/88fa9b6) New way to handle Streams
+* [(38d4312)](https://github.com/versatica/JsSIP/commit/38d4312) Add Travis CI support.
+* [(50d7bf1)](https://github.com/versatica/JsSIP/commit/50d7bf1) New `grunt grammar` task for automatically building customized Grammar.js and Grammar.min.js.
+* [(f19842b)](https://github.com/versatica/JsSIP/commit/f19842b) Fix #60, #61. Add optional parameters to ua.contact.toString(). Thanks @ibc
+* [(8f5acb1)](https://github.com/versatica/JsSIP/commit/8f5acb1) Enhance self contact handling
+* [(5e7d815)](https://github.com/versatica/JsSIP/commit/5e7d815) Fix. ACK was being replied when not pointing to us. Thanks @saghul
+* [(1ab6df3)](https://github.com/versatica/JsSIP/commit/1ab6df3) New method JsSIP.NameAddrHeader.parse() which returns a JsSIP.NameAddrHeader instance.
+* [(a7b69b8)](https://github.com/versatica/JsSIP/commit/a7b69b8) Use a random user in the UA's contact.
+* [(f67872b)](https://github.com/versatica/JsSIP/commit/f67872b) Extend the use of the 'options' argument
+* [(360c946)](https://github.com/versatica/JsSIP/commit/360c946) Test units for URI and NameAddrHeader classes.
+* [(826ce12)](https://github.com/versatica/JsSIP/commit/826ce12) Improvements and some bug fixes in URI and NameAddrHeader classes.
+* [(e385840)](https://github.com/versatica/JsSIP/commit/e385840) Make JsSIP.URI and JsSIP.NameAddrHeader more robust.
+* [(b0603e3)](https://github.com/versatica/JsSIP/commit/b0603e3) Separate qunitjs tests with and without WebRTC. Make "grunt test" to run "grunt testNoWebRTC".
+* [(659c331)](https://github.com/versatica/JsSIP/commit/659c331) New way to handle InvalidTargetErorr and WebRtcNotSupportedError
+* [(d3bc91a)](https://github.com/versatica/JsSIP/commit/d3bc91a) Don't run qunit task by default (instead require "grunt test").
+* [(e593396)](https://github.com/versatica/JsSIP/commit/e593396) Added qunitjs based test unit (for now a parser test) and integrate it in grunt.js.
+* [(da58bff)](https://github.com/versatica/JsSIP/commit/da58bff) Enhance URI and NameAddrHeader
+* [(df6dd98)](https://github.com/versatica/JsSIP/commit/df6dd98) Automate qunit tests into grunt process
+* [(babc331)](https://github.com/versatica/JsSIP/commit/babc331) Fix. Accept multiple headers with same hader name in SIP URI.
+* [(716d164)](https://github.com/versatica/JsSIP/commit/716d164) Pass full multi-header header fields to the grammar
+* [(2e18a6b)](https://github.com/versatica/JsSIP/commit/2e18a6b) Fix contact match in 200 response to REGISTER
+* [(3f7b02f)](https://github.com/versatica/JsSIP/commit/3f7b02f) Fix stun_host grammar rule.
+* [(7867baf)](https://github.com/versatica/JsSIP/commit/7867baf) Allow using a JsSIP.URI instance everywhere specting a destination.
+* [(a370c78)](https://github.com/versatica/JsSIP/commit/a370c78) Fix 'maddr' and 'method' URI parameters handling
+* [(537d2f2)](https://github.com/versatica/JsSIP/commit/537d2f2) Give some love to "console.log|warn|info" messages missing the JsSIP class/module prefix.
+* [(8cb6963)](https://github.com/versatica/JsSIP/commit/8cb6963) In case null, emptry string, undefined or NaN is passed as parameter value then its default value is applied. Also print to console the processed value of all the parameters after validating them.
+* [(f306d3c)](https://github.com/versatica/JsSIP/commit/f306d3c) hack_ip_in_contact now generates a IP in the range of Test-Net as stated in RFC 5735 (192.0.2.0/24).
+* [(528d989)](https://github.com/versatica/JsSIP/commit/528d989) Add DTMF feature
+* [(777a48f)](https://github.com/versatica/JsSIP/commit/777a48f) Change API methods to make use of generic 'options' argument
+* [(3a6971d)](https://github.com/versatica/JsSIP/commit/3a6971d) Fix #26. Fire 'unregistered' event correctly.
+* [(5616837)](https://github.com/versatica/JsSIP/commit/5616837) Rename 'outbound_proxy_set' parameter by 'ws_servers'
+* [(37fe9f4)](https://github.com/versatica/JsSIP/commit/37fe9f4) Fix #54. Allow configuration.uri username start with 'sip'
+* [(a612987)](https://github.com/versatica/JsSIP/commit/a612987) Add 'stun_servers' and 'turn_servers' configuration parameters
+* [(9fad09b)](https://github.com/versatica/JsSIP/commit/9fad09b) Add JsSIP.URI and JsSIP.NameAddrHeader classes
+* [(f35376a)](https://github.com/versatica/JsSIP/commit/f35376a) Add 'Content-Length' header to every SIP response
+* [(3081a21)](https://github.com/versatica/JsSIP/commit/3081a21) Enhance 'generic_param' grammar rule
+* [(e589002)](https://github.com/versatica/JsSIP/commit/e589002) Fix. Allow case-insentivity in SIP grammar, when corresponds
+* [(aec55a2)](https://github.com/versatica/JsSIP/commit/aec55a2) Enhance transport error handling
+* [(d0dbde3)](https://github.com/versatica/JsSIP/commit/d0dbde3) New stun_servers and turn_servers parameters
+* [(47cdb66)](https://github.com/versatica/JsSIP/commit/47cdb66) Add 'extraHeaders' parameter to UA.register() and UA.unregister() methods
+* [(69fbdbd)](https://github.com/versatica/JsSIP/commit/69fbdbd) Enhance in-dialog request management
+* [(da23790)](https://github.com/versatica/JsSIP/commit/da23790) Fix 'UTF8-NONASCII' grammar rule
+* [(3f86b94)](https://github.com/versatica/JsSIP/commit/3f86b94) Require a single grunt task for packaging
+* [(81595be)](https://github.com/versatica/JsSIP/commit/81595be) Add some log lines into sanity check code for clarity
+* [(a8a7627)](https://github.com/versatica/JsSIP/commit/a8a7627) Enhance RTCPeerconnection SDP error handling. Thanks @ibc for reporting.
+* [(3acc474)](https://github.com/versatica/JsSIP/commit/3acc474) Add turn configuration parameters for RTCPeerConnection
+* [(9fccaf5)](https://github.com/versatica/JsSIP/commit/9fccaf5) Enhance 'boolean' comparison
+* [(24fcdbb)](https://github.com/versatica/JsSIP/commit/24fcdbb) Make preloaded Route header optional.
+* [(defeabe)](https://github.com/versatica/JsSIP/commit/defeabe) Automatic connection recovery.
+* [(a45293b)](https://github.com/versatica/JsSIP/commit/a45293b) Improve reply() method.
+* [(f05795b)](https://github.com/versatica/JsSIP/commit/f05795b) Fix. Prevent outgoing CANCEL messages from being authenticated
+* [(5ed6122)](https://github.com/versatica/JsSIP/commit/5ed6122) Update credentials with the new authorization upon 401/407 reception
+* [(2c9a310)](https://github.com/versatica/JsSIP/commit/2c9a310) Do not allow reject-ing a Message or Session with an incorrect status code
+* [(35e5874)](https://github.com/versatica/JsSIP/commit/35e5874) Make optional the reason phrase when reply-ing
+* [(85ca354)](https://github.com/versatica/JsSIP/commit/85ca354) Implement credential reuse
+* [(351ca06)](https://github.com/versatica/JsSIP/commit/351ca06) Fix Contact header aggregation for incoming messages
+* [(d6428e7)](https://github.com/versatica/JsSIP/commit/d6428e7) Fire UA 'newMessage' event for incoming MESSAGE requests regardless they are out of dialog or in-dialog.
+* [(1ab3423)](https://github.com/versatica/JsSIP/commit/1ab3423) Intelligent 'Allow' header field value. Do not set a method in the 'Allow' header field if its corresponding event is not defined or has zero listeners.
+* [(4e70a25)](https://github.com/versatica/JsSIP/commit/4e70a25) Allow 'text/plain' and 'text/html' content types for incoming SIP MESSAGE Fixed incoming SIP MESSAGE processing when the Content-Type header contains parameters
+* [(d5f3432)](https://github.com/versatica/JsSIP/commit/d5f3432) Fixed the message header split when a parsing error occurs. Parsing error log enhanced.
+
 Version 0.2.1 (released in 2012-11-15)
 -------------------------------
 
diff --git a/README.md b/README.md
@@ -32,26 +32,50 @@ coolPhone.start();
 
 // Make an audio/video call:
 
-var useAudio = true;
-var useVideo = true;
-
+// HTML5 <video> elements in which local and remote video will be shown
 var views = {
-  'selfView':   document.getElementById('my-video'),
+  'selfView': document.getElementById('my-video'),
   'remoteView': document.getElementById('peer-video')
 };
 
+// Register callbacks to desired call events
 var eventHandlers = {
-  'connecting': function(e){ // Your code here },
-  'progress':   function(e){ // Your code here },
-  'failed':     function(e){ // Your code here },
-  'started':    function(e){ // Your code here },
-  'ended':      function(e){ // Your code here }
+  'progress': function(e){
+    console.log('call is in progress');
+  },
+  'failed': function(e){
+    console.log('call failed with cause: '+ e.data.cause);
+  },
+  'started': function(e){
+    var rtcSession = e.data.session;
+
+    console.log('call started');
+
+    // Attach local stream to selfView
+    if (rtcSession.getLocalStreams().length > 0) {
+      selfView.src = window.URL.createObjectURL(rtcSession.getLocalStreams()[0]);
+    }
+
+    // Attach remote stream to remoteView
+    if (rtcSession.getRemoteStreams().length > 0) {
+      remoteView.src = window.URL.createObjectURL(rtcSession.getRemoteStreams()[0]);
+    }
+  },
+  'ended': function(e){
+    console.log('call ended with cause: '+ e.data.cause);
+  }
+};
+
+var options = {
+  'eventHandlers': eventHandlers,
+  'userMediaTypes': {'audio': true, 'video': true}
 };
 
-coolPhone.call('sip:bob@example.com', useAudio, useVideo, eventHandlers, views);
+
+coolPhone.call('sip:bob@example.com', options);
 ```
 
-Want to see more? Check the full [Getting Started](http://jssip.net/documentation/0.2.x/getting_started/) section in the project website.
+Want to see more? Check the full [Getting Started](http://jssip.net/documentation/0.3.x/getting_started/) section in the project website.
 
 
 ## Online Demo
@@ -73,14 +97,20 @@ Check our **Tryit JsSIP** online demo:
 
 ## Authors
 
-### Main Author
+### José Luis Millán
+
+* Main author. Core Designer and Developer.
+* <jmillan@aliax.net> (Github [@jmillan](https://github.com/jmillan))
+
+### Iñaki Baz Castillo
 
-* José Luis Millán (<jmillan@aliax.net> | [github](https://github.com/jmillan) | [twitter](https://twitter.com/jomivi))
+* Core Designer and Developer.
+* <ibc@aliax.net> (Github [@ibc](https://github.com/ibc))
 
-### Contributors
+### Saúl Ibarra Corretgé
 
-* Iñaki Baz Castillo (<ibc@aliax.net> | [github](https://github.com/ibc) | [twitter](https://twitter.com/ibc_tw))
-* Saúl Ibarra Corretgé (<saghul@gmail.com> | [github](https://github.com/saghul) | [twitter](https://twitter.com/saghul))
+* Core Designer.
+* <saghul@gmail.com> (Github [@saghul](https://github.com/saghul))
 
 
 ## License
