My question isn't OS specific but I'll give you the details here just in case:
Server=Linux CentOS 5 with VLC 0.9.8a
Client=WinXP with VLC 0.8.6c
On the client I have an 8kHz mono 16 bit WAV file which I've proved plays correctly from the server's own soundcard using VLC's GUI. Hitting CTRL-I while the file is playing gives me this:
Code: Select all
-stream 0
|-codec: araw
|-Language:
|-Type: Audio
|-Channels: 1
|-Sample rate: 8000 Hz
|-Bits per sample: 16
|-Bitrate: 128 kb/s
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cvlc file.wav --sout '#transcode{acodec=ulaw,rate=8000}:duplicate{dst=rtp{dst=192,168.0.2,port-audio=1234}}'
I have Wireshark running on the client and can see RTP packets arriving correctly. If I expand the "Real Time Transport Protocol" node of a received packet I can see:
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Real-Time Transport Protocol
[Stream setup by SDP (frame 2)]
10.. .... = Version: RFC 1889 Version (2)
..0. .... = Padding: False
...0 .... = Extension: False
.... 0000 = Contributing source identifiers count: 0
0... .... = Marker: False
Payload type: ITU-T G.711 PCMA (8)
Sequence number: 3
[Extended sequence number: 65539]
Timestamp: 480
Synchronization source identifier: 0xda3afed0 (3661299408)
Payload: (long hex value)
Any ideas? (I'll quickly point out that I've successfully streamed MP3 files using this setup; only the ulaw 8kHz attempt results in silence).