Question

Audio streaming TCP/IP TCP_NO_DELAY ACK Delphi / Droopy Eyes / Indy 10.0 GSM CODEC 6.10 PACKET DELAY SCRAMBLED AUDIO

Asked by: TDDatATS

Hi all experts,

First I'll try to explain what our software is doing and then I'll give a description of the problem.

We have three applications :
- one application is reading analog audio from the audio card and converts it to an audio stream using the GSM 6.10 CODEC. The delphi component used for doing this is droopy eyes. After the conversion to the stream is done, the application sends this stream out to another application using UDP
- the second application receives the stream and sends it out via TCP/IP to a number of connected clients. No audio components are used here, only the Indy 9.0 Delphi components for read the multicast and send it out over TCP/IP. This application is in fact the TCP server.
- the third application (client application) receives the audio signal via a TCP/IP client and delivers the stream packet to the Droopy Eyes component who convert the stream back to analog audio.

As you al may know, TCP/IP is an ACK/NACK protocol causing one packet to be send, an ACK send from the client application and then sending the next packet. So, TCP/IP always waits for sending a new packet untill an ACK is received from the previous sent packet.
There is an option in the TCP layer TCP_NO_DELAY which allows tcp/ip to not waiting an ACK before sending the next packet.
It is necessary to set this option because the packets need to go out fluently to have a good steady audio connection.

Now following problems occur :
- sometimes a client with a bad connection did not ack the last x (can be 25 for example) packets yet causing the TCP server to halt and wait untill he receives ACKs again. This causes all clients to not have an audio signal for some time since the TCP server is completeley blocking
- Sometimes the audio signal is scrambled and it is not clear in any way anymore. This seems to be because several packets are stuck together causing the Droopy Eyes buffer to generate a too big load and not processing the packet.

My questions :

- is there any way that we can infuence TCP/IP to not wait on the ACKs so our server application is not blocking
- can we set an option so packets are NEVER stuck together
- and off course any advice on any level is welcome


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Asked On
2006-10-12 at 04:23:57ID22021857
Tags

delphi

,

audio

,

streaming

Topic

Voice Over IP

Participating Experts
1
Points
500
Comments
5

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Answers

 

by: grbladesPosted on 2006-10-12 at 04:44:34ID: 17714303

> - is there any way that we can infuence TCP/IP to not wait on the ACKs so our server application is not blocking
It is not possible to turn off the waiting of ACK's since this would require an infinite amount of memory in order to store the outbound data incase a retransmission is required. However you can increase the buffer size so that more data can be sent before an ACK is received. This is known as the TCP Window and there is also a feature called TCP Sliding Window which measures latency and packet loss and adjusts the window size automatically. For more info have a look at http://www.tcpipguide.com/free/t_TCPWindowSizeAdjustmentandFlowControl.htm

> - can we set an option so packets are NEVER stuck together
Packets will never get stuck. It is probably a bug in the droopy eyes software.

> - and off course any advice on any level is welcome
It sounds as if you should be using the UDP protocol instead of TCP.

 

by: TDDatATSPosted on 2006-10-12 at 05:12:31ID: 17714442

Thank you for your comments.

The problem of stuck together is a mistake of me. What I meant is that packets are sticked together.
Exampke, server sends packet 1, packet 2 and packet 3.
These come in at the client side as one packet. My question is, can we set something so packets are never sticked together and always come in as packet 1 , packet 2 and packet 3.

UDP is indeed better for this. We have some projects running smooth with this technique.
However, our client insisted that TCP/IP is used so no firewall configuration like NAT, ... is necessary....

 

by: grbladesPosted on 2006-10-12 at 05:30:41ID: 17714583

All you can do is set the PSH control bit in the application to try and send the data as soon as possible which will tend to cause lots of smaller packets being sent instead of one larger one. However this does not force the packets to be sent independantly. If the transmission halts due to too many unacknowledged packets then the data which is then buffered from the application will be sent in larger sized packets which I am guessing is the cause of your problems.
You could decrease the MTU of the network interface on the server as this will limit the maximum size of the IP packet which can be transmitted and hence ca cause fewer application pacets to be sent inside the IP packet in this situation.

http://tcpipguide.com/free/t_TCPImmediateDataTransferPushFunction-2.htm

 

by: TDDatATSPosted on 2006-10-16 at 00:30:45ID: 17736922

HI guys,

taking into accounts all your remarks which were very usefull, I think we'll have to agree that audio over TCP/IP is not realistic.
It is, if you allow big delays (like streaming radio's) or if you just use it point-to-point.
However, if you want to use it like we do (sending out small packets (65 bytes) every 45 milliseconds), will cause the application to fail whenever there is a client with a bad connection.

So, we're converting it to UDP with dynamic ports so no configuration is needed.  

Thanks all for the help

20120131-EE-VQP-002

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