Jitter Buffer in sipXMedia

[Nauman Sulaiman]

Hi, 

We are using the latest recon code on a number of platforms. We would like to know how stable 

the jitter buffer implementation is. We have found outbound RTP audio is clear at remote end.

Inbound audio is also good if jitter on the network is low. In poor network conditions we have noticed

for example on an Android handset that inbound audio replicates the audio on the 'line'. We captured 

the RTP packets and also file logged audio after MprDecode and compared and they are almost identical

so it seems no real jitter buffering is being done.

Does the jitter buffer implementation work? If so is there anything that needs to be done to adjust it etc?

Currently we are just using the code out of the box.

Thanks 

[Daniel Petrie]

Hi Nauman:

The open source sipXmediaLib jitter buffer is very simple.  It has a fixed length buffer and if jitter exceeds the buffer size, it will have drop outs.

There are commercial pluggins for the sipXmediaLib that provide high quality audio improvements to the audio by adding robust algorithms for AGC, VAD, adaptive dejittter and packet loss compensation.  SIPez provides a commercial plugin for sipXmediaLib:

http://sipez.com/sme.html

SIPez also offers a commerical Android toolkit which provides a JNI interface to ease Java based application development of SIP applications.

http://sipez.com/ezAndroid.html

If you would like more information about the commercial solutions, contact me directly.

Cheers,

Dan

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From: Nauman Sulaiman <nauma...@gmail.com>
To: si...@googlegroups.com
Sent: Wednesday, April 18, 2012 10:35 AM
Subject: [sipX UA] Jitter Buffer in sipXMedia