= Troubleshooting Sound Problems - pjsip.org =

Troubleshooting problems in the audio quality is one of the most difficult task to do with PJMEDIA, because mostly it is system specific. This page will try to provide a step-by-step troubleshooting guide to assist solving such problems.

== Understanding Media Flow ==

To better understand how audio media flows between media components in a call, lets use the following diagram:

http://www.pjsip.org/images/media-flow.jpg



== Types of Problems ==

There are tooo many audio quality problems than those mentioned below, but anyway lets try to enumerate them and present with some kind of troubleshooting guide:

=== No audio is heard on local speaker during the call ===



=== No Audio is Heard by Remote Party ===

=== Loud Static Noise ===

=== Brief Drop-Outs in Audio ===

One mailing list member reported this problem in this thread:
 http://www.pjsip.org/pipermail/pjsip/2006-November/001111.html

Quoting his email:
> Does anyone else experience brief dropouts in audio? I can easily recreate this by playing a wave file to a channel that sends it to a local SIP server that just echos the result back. It's almost impossible to go more than a few seconds without extensive glitching and pops as the sound cuts out only to come back.

[[BR]]

The solution that worked for him is to run the application in Release mode. Quoting his email again:
> Seems one shouldn't use pjsip compiled in debug mode. (-g.) Going  back to optimized cleared things up.

=== I'm sending tone in .WAV file from pjsua but got "stutters" on the remote side ===

One mailing list member tried to stream a .WAV file containing tone to X-Lite and SJPhone and observed audio "stutters" in the receiving side. But this didn't happen when the receiving side is another pjsua.

Solution: don't send tone file to these user agents, as it's suspected that they try to do something with the in-band tone. Use .WAV file containing usual speech and it should be okay.



=== High jitter value observed by remote party ===