Release Notes
Jan 16, 2012
- Moved functions from md5.pas into iputils.pas. Deleted md5.pas and also removed WSock2.pas from the project.
Jul 8, 2011
- When calling via firewall sometimes it did not recognize that the remote user hang up. This has been fixed.
Apr 26, 2011
- The component can make calls using account information (ISipAccount) without registering it on SIP server in case you don't want to receive incoming calls.
- If you don't need some codecs to be used in the system you may just remove it from the project. When you include the codec (like codecgsm.pas) in the project it registers itself in the "initialization" section of the unit and may be used in VoIP conversations.
- Removed rtp.pas and logmessage.pas. Added pulse.pas - virtual microphone for IVR systems, redirector.pas - implements ISipSpeaker interface which is also acting as a microphone for another ISipCall, so the incoming audio stream from one user may be redirected to another remote user.
Apr 12, 2011
- Added the ability to combine several .wav files and play at once, like SipClient1.PlayWavFile(Call, 'c:\wav\ninty.wav,c:\wav\five.wav');, which will play "ninty five".
Apr 11, 2011
- Soundinput.pas renamed to microphone.pas, soundplay.pas renamed to speaker.pas and RTP logic moved from these modules to call.pas.
- In some cases it used 127.0.0.1 as a local IP address. This has been fixed. Thanks to Markus Kollmann!
- Jitter handling logic has been changed in order to improve the sound quality.
- Reduced the load of the time critical speaker thread by moving the audio conversion to another thread.
Feb 9, 2011
The component did not handle correctly the incoming calls when all codecs sent in the INVITE message were not supported. This has been fixed. Now it's sending the error message 488 according to RFC 3261.
Jan 31, 2011
The property ISipAccount.DisplayName renamed to ISipAccount.Name. This property is used as "friendly name" in "from" address part of SIP messages. For example:
...
From: "friendly name"
To:
...
Jan 18, 2011
Added new property ISipCall.Reason type String. When your application receives the event OnBye which means the end of phone call, this property will contain the error code. For example 200 - OK, 486 - Busy here, 487 - Request terminated, etc.
Jan 4, 2011
Added new method TSipClient.SetCodecs(const Codecs: array of Integer) which allows users to set the list of codecs and its priority.
Example:
MySipClient.SetCodecs([3, 0]); // this will instruct the SIP engine to try using GSM 6.10 first and then PCMU if the remote user does not support GSM 6.10.