G723_RTP decoding from rtp stream problem
HI,
I've managed to establish rtp connection and send the audio stream in G732 format. The stream is ok, but I can't decode the response from the asterisk server. I use the same way to set the audio format for both sending and receiving:
TrackControl[] tcs = processor.getTrackControls();
if(tcs.length == 0) {
return; //should not enter this case
TrackControl tc = tcs[0];//we expect one channel only
String codecType = SoftPhoneSettings.audio_codec;
tc.setFormat(new AudioFormat(AudioFormat.G723_RTP, 8000.0, 16, 1, -1, -1, 192, -1.0, null));
tc.setEnabled(true);If I use the gsm_rtp or ulaw_rtp codec it works just fine on both sides (also decoded right by me), but with G723_RTP I here no voice. And also I got EXCEPTION_ACCESS_VIOLATION on processor.close();, which is also only if using this codec.
The asterisk server is working fine, but I can't figure out what exactly is the problem here, and why I send it correctly but I can't decode the response. I've also tried with setCodecChain, but the situation is the same. Can anybody help me with this, because I can't find the source of the problem and more important I still can't solve it. 10x in advance.
Is there any standard way for retrieving the G723_RTP stream, like using a player or?
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Pls help if anyone can ThanksDear andreyvk ,
I've read your post
http://forum.java.sun.com/thread.jspa?threadID=785134&tstart=165
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Nico from Italy -
Using RTP stream as data source
I have successfully written a program that reads G.711 audio from a wav file and sends it out over the network.
I'm trying to modify that program so that instead of getting its source audio from a file, it instead receives the audio from an RTP stream. So, in effect, it is simply receiving audio from one socket and sending it back out another.
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[java] javax.media.NoProcessorException: Cannot find a Processor for: rtp://172.30.18.140:32916
[java] at javax.media.Manager.createProcessorForContent(Manager.java:1663)
[java] at javax.media.Manager.createProcessor(Manager.java:627)
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<removed funky formatting><br>
Message was edited by:
fehHi,
I Had a very similar problem, but in addiction I had to store data in a buffer before retransmission. I resolved it employing low level classes. At this time, I still do not know any other solution, and nobody helped me in any way, also in this forum...
So, first you have to access single frames of the stream (Buffer or ExtBuffer objects) using a RawBufferParser, then you have to reconstruct a DataSource by multiplexing the frames (use an appropriate Multiplexer class, such as RTPSynchBufferMux).
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I have not tried to retransmit data "straightforward", without access single frames, because that was not my task. Maybe if you use RTPManager, it is possible to get a DataSource by the ReceiveStream object, and then to pass it to another RTPManager to create a SendStream.
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Alberto M. (Italy) -
Hi guys,
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How can I transmit a YUV file? How can I create the DataSource which has this YUV file without having problems for the further creation of processor etc?
Thanks in advance guys,
ChrisOk, I think I know what was confusing me. Apparently the setRate() isn't effecting the rate of stream. The audio file I am streaming runs about five minutes usually, so I sped the rate up to 10. It sped the process of writing to a file up significantly, but writing to an RTP stream seems to take the full run-time of the file. Does this sound right or is there something else I'm missing?
Thanks,
Khanathor -
Dear Java-Users,
we like to build a little RTP-Streaming radio.
We found a lot of code samples for example the AVTransmit/Recieve from java.sun.
Our client is working well and is able to recieve RTP-Streams.
But our server isn't running at the moment.
When we start both (server and client) on one local machine, the client can recieve the Stream. When we start the server on an other machine from the same network, the client cannot recieve the stream.
Well there are two problems left :
1) As I said the server does not work except from localhost.
2) We read that there is no MP3 support on Java-RTP-Streaming. But as you know MP3 is the most used audio-format and that's why we like to implement it.
If somebody could help me, I would be really glad.
Please send me an email or post here. ([email protected])
I appreciate for your time. with good greetings,
Tobias BelchReally I have to pay to use MP3 ?
No I didn't knew this.
I want to use MP3, for I want to build a simple and small client/server for RTP-Streaming over a standard network. And MP3 is the most used format and that's why it is just comfortable for the user to use mp3.
But thank you very much for your hint.
greetings -
Hello
I need help can someone tell me how i play and save at the same time an rtp stream?
ThanksThanks for your response.
Very much appreciated. Was very informative.
This is my current situation with 3 your suggestions:
Daniele,
Your suggestion 1’s result:
In Wireshark ----> Under Statistics --->I have VoIP calls.
(I don’t see VoIP calls under Telephony –> may be a different version of Wireshark).
Anyway, there is only one call because the Wireshark had a Capture Filter to track information between one source and one destination IP address. So I select that call and click on Player button and then click on Decode button. Then I select the forward stream (From IP1 to IP2) and click on play and I don’t hear anything at all. All silence. Same when I select the reverse stream from IP2 to IP1 and play.
Your suggestion 2’s result:
In Wireshark ---> Under Statistics ---> I Selected Stream Analysis (Did not select Show All Streams – not sure what the difference is) then ---> Save Payload ----> Select “au” instead of raw and it says – “Can’t save in a file:saving in au format supported only for alaw / ulaw stream
Your suggestion 3’s result:
Saved the file in .raw format. Opened Audacity and imported the file as raw and specified FIRST the A-Law codec for G.711A and selected 8000hz and that didn’t work and SECOND tried the u-Law coding for G.711u and selected the sample frequency again equal to 8000 Hz and that didn't work.
Didn't work means:
When I played the imported information I get all noise (like heavy metallic sound) and no voice.
So my guess is that this capture is neither A-Law or u-Law codec - right. This capture was given to me by a customer.
Any other suggestions – much appreciated Daniele.
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