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hmm.. I have set the hz/bit for both recorder and player in the same value, but the sound didn't play correctly, maybe do you have any experience for my problem?
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Then there can be only timing issues. You don't play audio samples with right rate(ms).
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sorry to bother you again Mr. Ivan. But I want to know, why there are so much noise? The noise almost covered all the sound. How to clear the noise or at least, the noise is much more quiet than my voice..
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If you are sending through netowrk, fisrt you may try send locally. This shows if network delay causes it.
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Actually, I just use this program to receive the sound from the mic and directly play the recorded sound in the same computer. If I use your program with using the same ip address for sender and receiver, the result is still the same. there are so much noise.
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hmm.. still the same. in the output the sound that heard was like a robot.
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Hi,
When I am using this code to encode/decode audio,
Sometimes there is significant delay in the audio reaching the listener.
For eg, someone says something on a microphone and it takes maybe 10 seconds for the listener at the other end to hear it...
Maybe its a network issue and there is nothing I can do about it but I was wondering if there was anything I could do to make this delay less.
Thanks
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This delay problem seems also be in RTP audio.
I have tested both versions WavePlayer and RTP Audio over a VPN with a roundtrip of about 42mS
and the delay is increasing over time, if i am connected for about 1 hour the delay can be up to 1.5 sek
sometime i have had several seconds in delay.
I have another voice application named IPSOUND that works OK, no delay what so ever, but that software is out of support and upgrades.
I had thoughts about making a new VOIP similar to IPSOUND that uses RTP audio, but first i have to sort out this delay problem.
Have you Ivar noticed any of this problem ?
Regards
Pete
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Hi,
>I had thoughts about making a new VOIP similar to IPSOUND
You can try:
Simple SIP (VOIP) based phone in C#[^]
You can use local IP instead of domain, you can call 1 phone to another.
Though you need 2 computers for that.
See if it works ok.
For WavePlayer and rtp audio, first you need to sort out if mic or play out causing delay.
There are no adaptive sound buffering, so i can't see how delay increases ... .
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Hi
I have downloaded Simple SIP and i am not shore how to set it up, i have 2 PC, one on 192.168.0.3 Local
and one remote 192.168.1.8
I have tried to set From sip:192.168.0.3 To sip:192.168.1.8 and on the remote side From sip:192.168.1.8 to sip:192.168.0.3 but i can´t get them to answer any call.
Am i missing something ?
>For WavePlayer and rtp audio, first you need to sort out if mic or play out causing delay.
>There are no adaptive sound buffering, so i can't see how delay increases ...
Well it works perfectly without any delay when i use IPSOUND with the same input / output so i don´t know.
Regards
Pete
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You need to use "sip:user@ip:5060" as To:.
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Hi Ivar
Strange, I cannot get the them to answer I have tried this settings.
Local machine
-------------
From sip:blue@192.168.0.3:5060
To sip:black@192.168.1.6:5060
Remote Machine
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From sip:black@192.168.1.6:5060
To sip:blue@192.168.0.3:5060
No one of them answer any call
I haven't look so mush at the code but are they listening to the From: IP when they start ?
I have tested to connect from both PC to echo@iptel,org and music@iptel.org and that is working
but they do not connect to each other as above.
//Pete
modified 23-Jan-12 16:50pm.
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Hi,
Do you get phone ringing ?
In one(which) computer you may leave all to from as is, on another you only need to alter to field:
sip:xxxx@remoteIP:5060
Phone listens on port 5060.
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Hi
Yes it ringing, text message says calling and i hear a tone every time, but there is no response on the other side.
Strange this, i have also tested on same network lan, ie 192.168.0.6 and 192.168.0.3, (not over VPN)
but there is no connection there either.
I have copied the files that are in the Bin/Debug to the other computer and the program run ok, on the other side i run it from visual studio, but i don´t think it can have something to do with that.
I will get up wireshark to se if i can se any IP traffic.
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I suspect the problems begin when multpile active network connections.
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I have look at wireshark and it seems to some network problem, it says Destination unreachable.
the packets are as follow.
Quote: 177 32.487319 192.168.0.3 192.168.0.6 SIP 429 Request: OPTIONS sip:ping@publicIP.com
Quote: 178 32.488903 192.168.0.6 192.168.0.3 ICMP 190 Destination unreachable (Port unreachable)
Quote: 180 32.897771 192.168.0.3 192.168.0.6 SIP/SDP 702 Request: INVITE sip:bl@192.168.0.6:5060, with session description
Quote: 181 32.899273 192.168.0.6 192.168.0.3 ICMP 190 Destination unreachable (Port unreachable)
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Hi,
I've been playing with your code and need to build a TCPServer to complete a demonstration I am making.
While browsing on your website (http://www.lumisoft.ee) I saw that you have a new version of LumiSoft.Net library.
I tried to use the compiled DLL with this project and it seems that you've moved things around and might have placed them as obsolete.
Is there a new version of your AdvancedWave sample with your new code?
Thanks,
Noam
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Thanks Ivar for the fast reply.
I played with it a little bit and got a few exceptions...
To do that simply start the server with your IP's (I used the local IP and 127.0.0.1) and start sending the mic data.
The sending microphone dialog box appears and then pressing start, waiting for a while then stop, then start again, waiting and stop will result with a few exceptions (or at least it was for me).
The error I got was :
Unhandled error: System.InvalidOperationException: Handle is not initialized.
at system.runtime.interopservices.GCHandle.AddOfPinnedObject()
at LumiSoft.Net.Media.AudioIn.WaveIn.<>c_DisplayClass1.<onwaveinproc>B__0(Object state) in E:\LumiSoft\Net\Net\Media\AudioIn.cs: line 693
at System.Threading._ThreadPoolWaitCallback.WaitCallback_Context(Object state)
at ....
Any suggestions?
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I dont know if helps, but i updated rtp demo to latest.
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If you did it today after 14:15 (UTC+2) then I should get the new version.
Otherwise....
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Hi,
I was just wondering if it was possible to achieve this functionality (of the encoding and decding) uisng NAAudio
http://naudio.codeplex.com
Thanks
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