I understand that we are adding functionalities only for Voip users, but
the code added for this will only be executed if necessary. I mean, it
is not part of main capture/dissectors (the Audio API will be
initialized/terminated in the "RTP analysis" only), the binary file will
grow and use more memory and time when Ethereal startup but that's all,
or I missing something here?.
Anyway, I'll start making some tests adding an external application to
do this job and see how it works and how big it is.
Regards
Alejandro
Lars Ruoff wrote:
Hi Alejandro,
the idea is great, but adding yet another multi-platform and highly
hardware-dependend library sounds risky in my opinion.
Also, don't we take the thing too far here for a protocol analyzer? Think of
the people that aren't involved in VoIP. Do they want to carry the audio
listener overhead?
So perhaps this should be done in a seperate application first? - to see if
it works well and how much space it will take.
The AudioListener app could work on rtpdump files as input, thus we have
easy interoperability with Ethereal but the app would be valuable well
beyond the scope of Ethereal.
(Then again, this might already exist to some extent...)
regards,
Lars
----- Original Message -----
From: "Alejandro Vaquero" <alejandrovaquero@xxxxxxxxx>
To: <ethereal-dev@xxxxxxxxxxxx>
Sent: Tuesday, April 12, 2005 3:58 AM
Subject: [Ethereal-dev] RTP audio listener
Hi All,
I'm planning to add an enhancement to the RTP analysis to listen the
audio. This is the idea:
1) Use an open source multi-platform audio library. PortAudio
(www.portaudio.com) looks easy to use and have all we need just to play
the audio, and works for Win, Mac, and unix (OSS). I definitely need
some help here to be able to add it to Ethereal and then test it in
different platforms.
2) There are codecs that need license to be used like G729 and G723. We
can not add this as part of Ethereal distribution, but the idea is to
allow Ethereal to easily use those as plugins if someone wants to add
them because it has license. So, the idea is to have plugins for the
codecs. We can start doing the G711 plugin. The plugin will take the
compressed audio buffer, and return the uncompressed audio (a 8khz,
16bit, mono)
3) Before playing the audio, will be a a variable to set the "jitter
buffer" parameter to simulate a real DSP. We can start using an static
jitter buffer, and then improve it to be dynamic.
Ideas/thought??
Regards
Alejandro
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