$Id: ChangLog,v 1.6 2000/08/12 01:23:55 wenqing Exp $ RTP ChangLog ---------------------------------- ---------------------------------- $Log: ChangLog,v $ Revision 1.6 2000/08/12 01:23:55 wenqing change for making new release Revision 1.5 2000/06/24 01:39:04 wenqing change for 0.1.22 Revision 1.4 2000/05/26 01:20:24 wenqing update for the new release Revision 1.3 2000/03/28 03:27:49 wenqing update for new release ---------------------------------- Rlease 0.1.23, w.r.t the previous releases - Change receiver to allow SSRC = 0. Rlease 0.1.22, w.r.t the previous releases - Chopped files to more files due to file too long - Fixed problem of late packets. Add handling to reorder the out-of-order packets. The lost packets are replaced by next packet available. - Fixed bug for sample rate/ pakect size issue and byte order for Linear 16 and other - Fixed/Enhanced for Sequence number checking and sequence number cycling - Add new tranmitRaw() method in RtpTransmitter.cxx to allow pass in raw data - Enhanced the half-duplex to full-duplex transmision - Fixed bugs for half-duplex mode - Fixed problem of UdpStack in util/. It's linux specific. The problem is a socket doign sendto() and recevfrom(), then get connection refused for the recv. Use setsockopt(...) to fix. - Other minor bug fixes. ---------------------------------- Rlease 0.1.21, w.r.t the previous releases - Fixed memory leak problem for both RTP and RTCP - Add more call statistics API in RtpSession, and correct the Jitter and Latency calculation - Upgrade test programs such as sampleUsage and phoneJack - Other bug fixes: such as one related to race condition and one in emptyNetwork(). -------------------------------- Rlease 0.1.20, w.r.t the previous releases - New test program called *phoneJack* for testing RTP directly with Quicknet card. - Improved test program/bug fixes for rptPlay/rtpRecord and adding explaination of how it works. - Fixes for one-way latency problem. This is because there is time difference between the two endpoints to start audio, and the stack doesn't do catchup at the beginning of audio. So whoever start's audio late will have latency problem. Now the stack will do catch-up when the audio starts. - Change from send() to sendto() and not doing *connect()* ports. So then the stack can receive RTP packets from other ports also can send to different port. - inactive/Sendonly/recvonly/sendrecv/ mode exchanges are OK now. -------------------------------- Release 0.1.19, w.r.t the previous releases - various bug fixes -------------------------------- Release 0.1.18, w.r.t the previous release - UDP stack, much enhanced. New object for network addresses. They have been moved to utility directory(util/). - Now can accept any packet size from network and output any desired packet size to API. - Bug fixes from previous releases - Naming changes for all the files which have class objects --------------------------------- Release before 0.1.18 SESSION IMPROVEMENTS - hooks for multiple CODEC and CODEC conversion - implement select() usage on receive calls - recovering from lost packets, reordering, and jitter - allowed dymanic port allocation within a given range RTP IMPROVEMENTS - correct network to host order for interoperability with other RTP stacks - fixed RTP timestamp so it relates to the sample rate RTCP IMPROVEMENTS - NTP timing corrected so stack is less clock dependent - corrected RTCP calculations