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Hello Ivar, I realize this is a long shot being that this post is so old but this looks like the best place to seek some help on this, it's been years since I've done any coding and I've forgotten most of it, I've been trying to get your program to work without a UI for the last couple days with no success.
I've been able to compile default ip addresses into local port and endport which works so that all my users have to do is click the 2 start buttons however I would like to get something working which doesnt require a user to actually do anything, what im trying to do is use either advanced wave or RTP Audio Demo or something similar without a user interface, I am trying to develop a program that I can startup on a remote system via TCP command prompt or some lower form of RDP, which will allow me to talk to my users while im troubleshooting their problems, most of my machines here have integrated mic's, so default values should work for most of the WaveIn and WaveOut classes.
Not sure how you'd select a new WaveIN/WaveOut device anyway with no GUI, unless there's a way to pull the current device being used out of a system property somewhere.
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Hi,
I dont see what problems you have, no UI is required to wave stuff. Also lisitng devices works, but you must programtically choose opne to use(because on Ui user can choose one from combobox).
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Can Anyone Provide me Audio Codec Code or DLL for C#. so that i can transfer the audio data with less size over the internet....
In this example u r using G711 which can reduce audio size almost half but i need something which can reduce the size of audio data more than half..... Please help me......
Regards
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When i try to transfer the audio data to other party using internet on TCP and make buffer at other end. so that i can play the data smoothly. but when we play the data. there is break in the audio playing and also audio got corrupted but data remains same, i have check that......
plz help me out........ what algo should i follow to play the buffer data...........with ur code.
Thanks
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TCP cant be used for audio transport, you should use UDP.
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i am also using UDP but still there is prob with audio playing speed......
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can u guide me how to play buffer audio data which play smoothly with your code(Advance wave)
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Hi
this all works fine. can u please tell me why i can hear the voice which i speak. there is no prob in coding. but can u tell me
what is prob of echo........ please help me out
Regards
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Hi,
You should use haedphones and mic.
Or otherwise you should use some echo cancelation algorythm.
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can u help me further me in it for echo cancelation.
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suppose i speak a word "Hello" at client A, i can hear "Hello" again after some same time at Client A...... can u please tell me ........
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too less explanation.........if we had the ability to write the code on our own why would we look for it online...
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Thank you. But what about license to use Lumisoft's libraries?
Can I use Lumisoft's libraries in my applications?
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Hi,
(license blelow: the main purpose of it is to protect my work, while allow evry one to use it free)
General usage terms:
*) If you use/redistribute compiled binary, there are no restrictions.
You can use it in any project, commercial and no-commercial.
*) It's allowed to complile source code parts to your application,
but then you may not rename class names and namespaces.
*) Anything is possible, if special agreement between LumiSoft.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED,
INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR
PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE
FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE,
ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
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Thank you! If you need anything from me, by way of saying thanks, please let me know!
-Brendan
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Hi Ivar,
is there allready a RTP-implementation of this nice application ready to test?
- Markus
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I see there is a 'SIP_RTP.zip' uploaded on 9/18/2008 in the list with downloads(on lumisoft.ee)? Is it the new version of your lumisoftnet with improved RTP? If it is, is the code ready to use?
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Ye this is total rewrite of old, currently only missing part is jitter buffer.
Any feedback would be welcome ... .
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It looks and works very nice
A few comments
In your example code you use:
packet.Timestamp = (uint)timeStamp;
...
timestamp += 25
BUT the timestamp is not used. When I take a look at what is sent (with Wireshark), it is the same as the squencenumber.
I found the following in the code of lumisoftnet :
buffer[offset++] = (byte)((m_SequenceNumber >> 24) & 0xFF);
buffer[offset++] = (byte)((m_SequenceNumber >> 16) & 0xFF);
buffer[offset++] = (byte)((m_SequenceNumber >> 8) & 0xFF);
buffer[offset++] = (byte)(m_SequenceNumber & 0xFF);
And.. what will happen if the sequencenumber has an 'overflow', is your code able to handle that? (Did not look at that part of the code)
modified on Monday, September 29, 2008 4:09 AM
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Thanks reporting, it's a bug. Now it's fixed i also updated downlaodable version.
Also updated timestamp stuff, +25 not right, RTP clock is no implemented and timestamp value must be aquired from there.
Sequnce number wont overflow, it's wrapped around(if uint exceeded) as RFC says.
Just jjitter buffer and RTP frame constructor must be coded.
Probably must be separate components, because RTP frames used by video payloads only, so audio doesn't need that overhead checks.
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There is a property called 'IsMaker', maybe it has to be 'IsMarker'? (It is a property of the RTP_Packet Class)
modified on Friday, October 31, 2008 7:20 AM
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Yes you are right - this is typo, i fix it.
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