SIP Application for Red5 1.0 http://red5phone.googlecode.com/svn/branches/red5_1.0_compatibility

You would need to upgrade to Flash Player 11.0 to use this version. It supports only the G711 ulaw codec available in Flash Player 11.0. Nellymoser encoding/decoding has been removed.

Older versions are still available here
http://red5phone.googlecode.com/files/sip_r54.zip.
http://red5phone.googlecode.com/files/sip_r53.zip

New project "red5sip" is available as a branch at "https://red5phone.googlecode.com/svn" See http://incubator.apache.org/openmeetings/red5sip-integration.html and https://cwiki.apache.org/OPENMEETINGS/red5sip.html for details

Unzip and move the sip folder to webapps.

You will need to install Red5 server 0.9.1 Run red5phone application with http://your_red5_server:5080/sip

To checkout SVN use http://red5phone.googlecode.com/svn/branches/improvements/sip/

Changelog

R63

Repackaged to avoid confusion and to allow sharing between standalone and red5 app versions. Created Maven pom. Made compatible with Red5 server 1.0.

R54

Removed Nellymoser codec and now only supports G711 (ulaw) for now. Flex client has been modified. The laszlo versions may not work. They were not modified, recompiled or tested.

R53

Flex softphone compiled for Flash Player 10.3 to use Enhanced Microphone with no echo

R47

Updated Asao codec. Decoder and encoder both working ok. Bandwidth is now 8k in both directions.

R42

Fixed audio delay by adding timestamp using OS clock.

R40

Compatible with 0.9.1. Fixed issue 89.

R30-R39

Improvements by Rafael

R29

Incoming call failt falt fixed by Marcin Balcer

R28

Timestamp fix by Rafael on RTPSender Ignore non-audio RTP packets for RTMP audio by Naoki

Transfer feature by Lior

Blind transfer button to be used as followed:

a. register to your proxy b. make a call or get a call (than the transfer button will show up after call is answered) c. change the text of the dialed number to the number you want to transfer the call too d. press transfer button.

Conference feature using transfer by Dele

a. set your conference number b. call person. click on conference button to put caller in conference c. do that for all participants and click on conference icon to join confeence

R27

Dele Olajide Made red5phone compatible with Red5 server 0.8.0

Lior Herman Added outbound proxy settings in flex object, to solve realm problems.

Fixed DTMF. Tested with UK PSTN and works ok. Added Nellymoser encoder from Joseph Artsimovich (jart@users.sourceforge.net).

Mauro Brasil, Rafael and UOL (www.uol.co.br) contributions

Cleaned up the code

Implemented multi-codec support for Red5phone (PCMU, PCMA, iLBC and G.729). G.729 requires payment of royalities and license rights if used.

Adjustments to support different packetization were made as well.

Prabhu Tamilarasan contributions

Normalize function to improve volume

Lior Herman contributions

Red5phone show busy or rejected message for busy outgoing calls.

Openlaszlo and Javascript API made compatible with latest chages to Red5phone Add Dial external command to Flex to dial from a link.

Used UID (unique identifier) in Flex instead of the username value for binding the sipprovider object in Mjsip. Now you can use same sip account to register from multiple remote locations.

Changed registration of red5phone using phone@sip_provider.getViaAddress() instead of phone@realm

Mjsip now uses Outboundproxy = Proxy like that all sip headers using realm, but the message is send to the proxy ip.

Process of OPTIONS messages in Mjsip some sbc’s using to check call keep alive.

Fix authentication header for REG and INVITE to use cnounce value when proxy send qop= auth in 401 or 407 message.

Fix red5phone application close for complete unregister before sipprovider.halt is execute.

Fix the microphone.init in Flex to not be bind to the REGISTER SUCCESS message as it can be many SUCCESS messages coming during a call and mic need to be init only once.

Fix Cancel message for Invite (call) that have not been answered yet. When you want hangup a call before the remote user answered Mjsip send wrong Cseq in Cancel message. Fixed NullException error when application is closed in provider.halt() its try to close a null tcp_socket and get Null exception back. Fix is in org.zoolu.net.TcpServer.java in the end of the file.

Janny_buh contribution

Patch all sip Register message using same Call-ID header and increment Cseq as recommended by RFC3261.