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Hello Ivar,
I have been testing your program and i have had problems sending the data over the network. In contrast i've received data from the other end point (not very clear audio, but understandable). Is there some part wrong or is this the behaviour of the program? Maybe i should wait for RTP audio? or you could give me some notes to this problem and go on in this way??
Thank you in advance
Sergio
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Hi,
If network speed is ok and latency not bigger than 50 ms, the voice quality should be as phone quality.
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Hi again,
Thank you for the quickly reply, but i think that this is not my problem because i've tested in my local network, also my internet connection is good. I've been watching the transmited packets between the two points and i don't send anything good. I think that i take always the same number of bytes, but not real bytes i should take from the microphone. Regards as the packet i send, it is not RTP? How is the structure the sended packet?
Sergio Garcia
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It's just raw wav data, encoded/decoded before sending/receiving.
You try to turn off your firewall.
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Hi Ivar,
Ok, i will try with this. I'll tell you the results.
Thanks
Sergio Garcia
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hi Ivar,
as per above subject, i am offered to develop an application whicch is able to send/receive audi video data over LAN.
A kind of Video chat. a person cud able to watch and hear another person sitting in the same network.
while goggling for this... i got ur this article. Can u guide me pls...
Thanx
IPS
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Hi,
The most wisest approach is to use RTP. You just do 1 RTP stream for video, another for audio.
Tough i havent completed RTP yet ... .
You off course you can reuse my code, you can do 2 streams too 1 for video another for audio.
Capturing video and playing it, i havne't looked how to do it, so far i don't have needed it.
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Hi, how if I want to combine that 2 streams into only 1 stream which contains both audio and video ? I'm afraid if I use 2 streams then if the audio packet arrives late then the audio will also play late and vice versa.
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Hi,
You need RTP for that.
You must create 2 rtp sessions, 1 for audio, other for video.
(Because RTP specification allows 1 session to exhange only 1 type of data at time)
RTP has "timestamp" for synchronizing streams like you need.
Though RTP just provides all info you need for sync stream, but streams sync must be done RTP consumer app.
Normally timestamp is NTP clock time.
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Hi, does your library include classes or functions which I can use to transmit video ? I see there is only audio for codec and there is no video codec. And is there any complete documentation of your library ? I may need it. Thanks.
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Currently no video support and also i'm currently rewriting RTP. RTP finished probably during week.
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its great i like it i was looking for something like this cooooooooooooool
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Hi Ivar,
At here,we're waiting for your RTP audio multicasting contribution,and also I'm one in that.
Bigbermusa
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Hi,
It will be available some day, currently running out of time, need todo some real job mean while, billls must be payed, then free stuff can be coded.
It wont be available before next month.
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Hi,
However,I think we're waiting for it and understand your precious time.
Bigbermusa
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Just for letting you know, I am waiting for your RTP version, I am going to work with an Intranet Chat, And thinking this is a good stuff for adding to it, If I use it, i will let you know. GREAT JOB, MAN
Bye
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Hi ivar,
Can you give me idea about implementing Echo cancellation, Noise Reduction, VAD ,Gain Control ,etc. on your example. I looked over internet , but I can not find any resources on .net(c#,vb.net). Also I am waiting for your Server based voip article. Thanks.
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Hi,
I dont have such examples now, but probably there will be, if i get RTP done.
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I tested the application and the voice quality is not good at all (hopefully there's a problem with my PC, mic, etc.).
Is the LumiSoft.NET, a good dll for voice recording and playing?
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Hi,
But what about test sounds ? Are these ok ?
And between hosts you send sound, whats ping time ?
So far i haven't seen any troubles, but probably some toher testers comments are very welcome here.
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Hi Ivar,
I love active authors that reply soon I wish you the best in whole aspects of your life.
By the way, I didn't tested sound cause I only need the voice but I'm completely confused today. I'm trying to run two instances of WavePlayer locally and record the incoming voice; two record files are saved one 0kB and the other more than zero. But when I try to play the recorded file nothing is heard!!
Actually, I'd like to be sure of the quality of this dll to start creating an RTP session as you said to implement the voice conference (I wish I had time to wait for your next article).
One more question:
Since I'm trying to use this in web application I wonder if there's anything required to be installed on client machine to be able to perform voice conference.
Sorry if I take your time much
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> I'm trying to run two instances of WavePlayer locally
You can use same instance too, because its 2 way, Start button makes applicatyion to listen incoming audio on the specified port. Below stuff is for sending, if you cick send mic or soud, sound will be sent to the specified host.
FOr eaxample, you listen on 127.0.0.1:11000 and send to 127.0.0.1:11000, then it will record and play sent autio through same instance. But offcourse you can use 2 instances too, but ensure that listen ports wont match !!!
>start creating an RTP session
Qood luck, probably you wont imagine complexity of RTP, i have messed with it over month, probably soon will get it completed.
>Since I'm trying to use this in web application I wonder if there's anything required to be >installed on client machine to be able to perform voice conference.
Running on web, probably you run into many securty blocks what IE does.
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Hey, I'm not going to start RTP session on my own and from scratch I'm just looking around for other helpful articles:
http://www.codeproject.com/useritems/Using_RTP_in_Multicasting.asp
http://www.supinfo-projects.com/fr/2004/rtc_api/2/
http://www.faqs.org/rfcs/rfc3550.html
I'm sure you can understand them much better than me, cause you're experienced.
By the way, do you have any estimation on when you'll be completing your article?
Anyways, Fadi has used TAPI3 for voice conferencing (http://www.codeproject.com/useritems/Video_Voice_Conferencing.asp?df=100&forumid=373359&exp=0&select=2195323&msg=2195323), I'd been digging in it for a while before finding your article. Now, I'm a bit confused of which one to use, since it's not possible for me to test any of them under overload of users, etc. . Yours has a priority over that cause you're accessible to resolve my problem but is that a good reason?
Besides, I didn't get you >Running on web, probably you run into many security blocks what IE does. You mean the security blocks prevent me to use the dll on clients?! if yes, it should be a point of concern in all voiceconferencing web appl.
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>By the way, do you have any estimation on when you'll be completing your article?
Probably during next month.
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Very nice article (Actually, a hope light in darkness for me )
I was looking for a way to implement voice conference over internet as followings:
I need to let many users connect to the server and hear the voice but only one user can speak at any time. I mean, each user can be put on a queue and speak for a few minutes and then the next user will speak.
Any idea is appreciated cause I really need to complete my project soon.
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