Janice Lau wrote:
1) I have a standard voice file .wav on hand and i would like to send
it to a destination port using rtpplay and thus capture the packets
using Ethereal. May i know is there any way/tool that can help me to
convert the .wav voice file to .rtp file with all the header
information attached?
rtpplay can only send a captured RTP stream. I don't believe it can
create a new stream from a pure audio file. You may find sample
programs at http://www.live.com/liveMedia or
http://www.gnu.org/software/ccrtp/ but some programming will probably be
involved. It's strange that no one has come up with a simple way to do
this (that I know of).
Keep in mind that .wav files are straight PCM audio data and would
normally be encoded as a G.711 stream. If you want to test other bit
rates (e.g., G.729), you'll need a codec to convert the .wav file before
sending it to an RTP layer.
2) or is there any way i can directly send the .wav voice file to a
specified UDP port and then capture the network traffic using
Ethereal?
No. Keep in mind that the "RT" stands for Real Time. Streaming audio
is not a simple concept. You need an application that can break down a
digitally-sampled sound into time-coded packets and send them at precise
intervals over the network. That's what ccRTP and other implementations do.
If you want something simple, it's probably easier to capture a
good-quality RTP stream and start with that. There are many other
forums and mailing lists dealing with media applications, also.
-Scott