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Release Notes

PortSIP VoIP SDK V7.1
=============================================

Jan 16, 2012

1. Added the AMR, AMR-WB, G.722.1 codec.
2. Added two functions: setAudioCodecParameter and setVideoCodecParameter.
3. Removed the setProfileLevelIdOfH264 function.
4. Support RFC4867.
5. Added audioPlayLoopbackTest function.
6. Fixed a few bugs.

 

PortSIP VoIP SDK V7.0
=============================================

Nov15, 2011
This is a major upgrades.


1. Optimized the AEC and CNG, NS features, now enjoy the crystal HD audio.
2. Optimized the video codecs, now enjoy the crystal HD video.
3. Optimized the log system.
4. Support trace error from return value of API functions.
5. The G.723.1, G.722.1, AMR-WB are temporarily unavailable, will come with next 7.1.
6. Changed the audio and video stream callback functions.
7. Integrated the "Device Manager APIs" into "PortSIP Core".
8. Deleted some functions and added some new functions.
9. Support record video file as H.264, H.263 format to reduce the recording file size.
10. Optimized the core algorithm to reduce the CPU usage.


PortSIP VoIP SDK V6.8
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June 10, 2011

1: Support IPv6.
2: Added three fucntions for conference:
createConferenceEx
joinToConference
removeFromConference

3: Added two functions to allow obtain the statistics of RTP.
getAudioRtpStatistics
getVideoRtpStatistics

4. Added a new function which is getLocalIP6 to obtain the local IP in IPv6 format.
5: Added a function which is setKeepAliveTime to allows enable/disable the SIP keep alive.
6: Fixed some minor bugs.

 

PortSIP VoIP SDK V6.6
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Mar 10, 2011

1: Added support QVGA resolution for H.264 codec.
2: Added a new function which is detectWmi to obtain the WMI status.
3: Fixed a bug for the G.722 codec.
4: Added a function which is sendOutofDialogMessage.
5: Added new function sendMessageEx and sendOutOfDialogMessageEx allow send the binary data by MESSAGE method.
6: Added a function which is discardAudio to discard incoming and outgoing audio packets.
7: Added support switch audio device during the call.

 

PortSIP VoIP SDK V6.5
=============================================
October 20, 2010

1: Added MP3 format support for conversation recording.
2: Support play the wave file to a special line(session).
3: Support record the conversation on a special line(session).
4: Optimized the RTP jitter buffer.


PortSIP VoIP SDK V6.3
=============================================
August 20, 2010

1: Added a parameter which named appLogLevel for initialize function allows generate the SDK log into files.
2: Change enableLog function to enableStackLog.
3: Fixed a bug of PortXMPP that to compatibles with OpenFire.
4: Fixed a bug when call to PSTN got the bad voice quality if the ptime more than 20.

 

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Technical Tip

Question How to get the best voice quality?
Answer – Most of mobile computers have an embedded microphone and speaker. Make sure that your microphone is good enough by trying to record your speech. In order to get better audio quality and avoid echo, you may use headset, Blue Tooth, and wired headset etc;

We strongly recommend you to install the latest version of operating system, sound and network drivers, and DirectX (check with dxdiag.exe, it should be 9.0 or higher).

Check your Internet connectivity by pinging your party's IP address. A proper response time is below 100 ms unless the party you are calling is from Down Under. If the ping time seems too large, you Internet connection is too busy now. Try to stop any other activity. If the problem persists, ask your Internet provider or system administrator to fix it.

For the best voice quality, use the G.711, G.722.1 codec on broadband, on dialup try the G.729, iLBC, or other codecs;

For more information, read the FAQ.



 


 
 

 

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