Hello,
I have written two programs, based on AVTransmit2.java and AVReceive.java from the Sun JMF Solutions web page and I have successfully managed to tranmit a low-quality audio file (dvi) across the network.
When I try to transmit an mpeg layer 2 audio file, the receiving program successfully picks up on the RTP stream and recognises the "mpegaudio/rtp" format, but the sample rate is unknown:
-------------------------------------------------------------------------------------------
- Recevied new RTP stream: mpegaudio/rtp, Unknown Sample Rate
The sender of this stream had yet to be identified.
- A new participant had just joined: root@server.victor.lan
- The previously unidentified stream
mpegaudio/rtp, Unknown Sample Rate
had now been identified as sent by: root@server.victor.lan
-------------------------------------------------------------------------------------------
The transmitting program can recognise the sample rate, so am I right in assuming that this extra detail being lost in conversion to RTP packets?
Any help would be most appreciated. The page at:
states, under the RTP Formats section, that mpeg layer 2 audio is currently supported by the jmf rtp api and is suitable for both transmission and reception. Please, if I am just being incredibly stupid and I'm missing something glaringly obvious, don't hesitate to point it out!
The server is running Slackware 9.1 with the JMF Linux Performance Pack 2.1.1e, the client is running Fedora Core 1, with the same JMF Linux Performance Pack
I have successfully streamed an MP2 file using the JMF RTP API to a client machine running Win ME with the Windows perfomance pack JMF installed.
Thanks you in advance.
Pete
I have written two programs, based on AVTransmit2.java and AVReceive.java from the Sun JMF Solutions web page and I have successfully managed to tranmit a low-quality audio file (dvi) across the network.
When I try to transmit an mpeg layer 2 audio file, the receiving program successfully picks up on the RTP stream and recognises the "mpegaudio/rtp" format, but the sample rate is unknown:
-------------------------------------------------------------------------------------------
- Recevied new RTP stream: mpegaudio/rtp, Unknown Sample Rate
The sender of this stream had yet to be identified.
- A new participant had just joined: root@server.victor.lan
- The previously unidentified stream
mpegaudio/rtp, Unknown Sample Rate
had now been identified as sent by: root@server.victor.lan
-------------------------------------------------------------------------------------------
The transmitting program can recognise the sample rate, so am I right in assuming that this extra detail being lost in conversion to RTP packets?
Any help would be most appreciated. The page at:
states, under the RTP Formats section, that mpeg layer 2 audio is currently supported by the jmf rtp api and is suitable for both transmission and reception. Please, if I am just being incredibly stupid and I'm missing something glaringly obvious, don't hesitate to point it out!
The server is running Slackware 9.1 with the JMF Linux Performance Pack 2.1.1e, the client is running Fedora Core 1, with the same JMF Linux Performance Pack
I have successfully streamed an MP2 file using the JMF RTP API to a client machine running Win ME with the Windows perfomance pack JMF installed.
Thanks you in advance.
Pete