Teamspeak delay

[TonyMont]

We use teamspeak for a long time now, it works great, only all those years there is 1 thing that make us crazy, the teamspeak delay, we did several test's, skype and ventrillo is much faster then teamspeak, almost no delay.

The thing is that we play wars online, we have a big multigaming clan, loads of members.
When i get sniped from a spot, and i want to tell the spot to my team m8 its most of the times too late. They always say, i heard it too late.
We now play very important wars on skype, but that is not realy a good sollution if the rest of the clan is still on teamspeak.
This delay is not realy acceptable when you are playing official wars.

Is there a way to make the delay MUCH smaller, like the small delay that skype and ventrillo got?

I think i speak for most clans that have the same problem with the delay.


Friendly regards,


TonyMontana

[Judas]

1) have a look at exactly this page
2) scroll to the top if required
3) find the "Serach Forums" field
4) type in "delay" and gently click on the button that does say "Go"
5) be patient for a short time
6) start reading

[TonyMont]

1) have a look at exactly this page
2) scroll to the top if required
3) find the "Serach Forums" field
4) type in "delay" and gently click on the button that does say "Go"
5) be patient for a short time
6) start reading

I did.

I dont see many topic's about this problem.
I always look for a solution before using other software.
Maybe the new version got an option that removes some delay, then i can wait.
Or maybe there is some other solution that only teamspeak experts know about.
I dont mind if that solution kost more bandwith.
But what i need is a working solution, we can wait some weeks before installing other voice program on our 20 gameservers, this delay is not realy acceptable for online gamers.

What i need is a proper answer, i put allot of time in posting the thread, and ofcourse i searched the forum, your answer does not help at all.


Tony

[Timeless]

A delay of 0.25 or 0.8 sec doesn´t matter in Clanwars, my experience.
You need much more time to decide if you want to tell someone, despised the time you need to decide what to tell.

If there´s a remarkable delay in TS anyway, would you give up the advantages of TS for fractions of a second??

Every voice programm has got small delays, it´s just a matter of codec.

Talking about Skype:
It uses much more bandwith, which has a bad effect on your gameping! unacceptable.

We (my Clan) play Bf2 in the famous european Clanbase 32 ladder, are the best german Team in that ladder (Top 10), if we lose, it´s not because of TS, but because of bad tactics or a better opponent.

[TonyMont]

A delay of 0.25 or 0.8 sec doesn´t matter in Clanwars, my experience.
You need much more time to decide if you want to tell someone, despised the time you need to decide what to tell.

If there´s a remarkable delay in TS anyway, would you give up the advantages of TS for fractions of a second??

Every voice programm has got small delays, it´s just a matter of codec.

Talking about Skype:
It uses much more bandwith, which has a bad effect on your gameping! unacceptable.

We (my Clan) play Bf2 in the famous european Clanbase 32 ladder, are the best german Team in that ladder (Top 10), if we lose, it´s not because of TS, but because of bad tactics or a better opponent.

Our ping is still acceptable with skype.
But there is a remarkable difference, not talking about ms but seconds. maybe 2 or 3 seconds later then skype.
Host is 100 mbit LAN on gigabit router of a proper host in the Netherlands.
Its looks like the problem is something else..., i have it on all my teamspeak servers.
We can try other codecs...
Are there some other options that we can disable for beter performance?


Tony

[Judas]

If you did spend time researching, how could all those posts explaining things in detail have eluded you?

http://forum.goteamspeak.com/showthread.php?t=31731 (this is just an example, there are quite a few more)

I don't use skype myself but appart from what i know, skype doesn't use a client server model which obviously is faster (but more annoying to setup when you are behind a firewall). This also requires every client to send data to ever other client which is ridiculous as soon as you have more than just a bunch of clients.

and if you will have a look at the developers blog (at least i think i've read something there), you'll even find out that the problem has been recognised and that they are trying everything to make communication even faster.

And last but not least, if you really did spend that much time reading other posts and trying to solve the problem yourself, it would have helped a hell lot if you'd have said anything like that, mentioned what you have actually tried and on what your assumption that there is a huge delay is based.

[SuperTyphoon]

There is an annoying ass delay in TS, it's like 3 seconds usually or sometimes worse on laggy servers.

[WolfStar76]

There is an annoying ass delay in TS, it's like 3 seconds usually or sometimes worse on laggy servers.

3 seconds?

Most of the servers I've been on (which is admittedly only about 5) have *maybe* a half-second delay.

[sgtbenc]

I don't use skype myself but appart from what i know, skype doesn't use a client server model which obviously is faster (but more annoying to setup when you are behind a firewall). This also requires every client to send data to ever other client which is ridiculous as soon as you have more than just a bunch of clients.
FYI: Actually, Skype does not require you to send your voice data to all other clients. When you start a conference call, all voice data is routed through you.

[XeaLouS]

is there a way to reduce delay by reducing encode time but increaseing packet size? i.e. do less compression so that data is sent instantaneously? useful for lan is what im thinking. Very annoying haveing a 500ms delay in LAN... in cs the microphone function has very little delay.. cant that codec be incorporated?

[Timeless]

is there a way to reduce delay by reducing encode time but increaseing packet size? i.e. do less compression so that data is sent instantaneously? useful for lan is what im thinking. Very annoying haveing a 500ms delay in LAN... in cs the microphone function has very little delay.. cant that codec be incorporated?

Interesting question

Some people say that a low bandwidth codec (GSM 16 or Speex 12) have a lower latency. I say it´s unproven till I see a trustful protocol with measured delays.

I think there´s no difference in latancy results using one codec (even 12 or 25 kBit/sec), but I think there are differences with different codecs in comparison (e.g. GSM or Speex).

As far as I know there´s no effect of a lower compression rate, cause the algorithm is the same.

Questioning the packet size:
I don´t know, maybe some of the dev have further information to this question.

[LuxNegra]

TonyMont , what you could try in case if you haven't already , is to :

- Go to settings
- Options
- And there put the : "direct sound buffer size" to lower latency , and not better quality.

I don't know if it helps you , but it helped me. All of us on our ts server , are using the same config and the latency is much better now.

[Gylthinel]

If you did spend time researching, how could all those posts explaining things in detail have eluded you?

And last but not least, if you really did spend that much time reading other posts and trying to solve the problem yourself, it would have helped a hell lot if you'd have said anything like that, mentioned what you have actually tried and on what your assumption that there is a huge delay is based.

Thanks for the help.

I have latency problems w/ my TS too. I'll try the solutions presented on this thread by those who didn't ride in here on their high horse. To those, I give my thanks. :)

[Peter]

Hey guys,

Voice latency is a tricky topic, as numerous factors play into the end-result, these factors are:

Your ping on the server and the ping of the guy you are talking to. As a bare minimum the voice data will take YOUR_PING_TIME + HIS_PING_TIME milliseconds to "travel" the network from you to him. You can view the ping in TS2 by requesting connection info's of the people you whish to know the ping of.
Buffers...Every VoiceCom needs to have some sort of buffering in place, just incase some voice packets arive "late" so there is enough time to catch up and play everything that is being said afterall. The size of these buffers basically determines both howlong the voice data will "sit" after being received before being played (adding latency), and also how resistant the transmitted voice data will be to connection fluctuations (e.g. you start a heavy download). In TeamSpeak2 the sizes (of some?) of the buffers are configurable.
Codec "Chunk-Size". Depending on Codec and also on the way the Codec is handled by the VoiceCom you package sound-data in "chunks" of x-bytes (or y milliseconds of sound, if you like that better). Bigger chunks mean less overhead when sending over the net, but also higher latency. So if for example we were to send 1second of encoded voice-data per "chunk" then the data is at least 1 second old when it arrives at it's destination, even after disregarding the other points I mention in my post. With TS2 GSM 16.4 sends 50ms packets, CELP 5.1 and CELP windows 5.2 200ms packets and all other (all Speex codecs, the other GSM and CELP codecs) at 100ms packets.
Soundcard Buffers. Well, Soundhardware does buffering too, so if you use cheap onboard soundcards this might be a cause for added in latency, but I don't think this point will make the killer difference (I don't have hard data at my hand at the moment though)


So, what can I do to get lower latency sound transmissions with TS2, I hear you ask?

Lower the ping of you and your buddies to the TS2 server. This can be done by getting decent connections to the internet, by disabling downloads/uploads that you might have running in the background. Also moving the server to a "smarter" place might make sense. If you guys are a clan from Australia (and all live down under) then it would be stupid to use a german TS2 server.
Go to sound setup, where (this is from memory so you might have to fill in the gaps) there are two settings "Wave" and "Direct Sound"...Direct Sound will give you a slider which you might want to try pulling to lowest latency, with Wave you will not have this slider, but you *should* have a a smaller delay than you can achieve with Direct Sound, even with lowest buffer settings [note wave can cause issues with poor soundcards though, see [4] also :P]
Try avoiding the CELP *5.x codecs, use speex or if you strive for perfection GSM 16.4, although I doubt those 50ms will change a thing...
Get a good soundcard and stop using that onboard crap.



Also note that TS2 will not be killer low-latency even after taking all these points into consideration, they are merely a way to improve the situation. Even more improvement can be made but it requires changing the way TeamSpeak works...see the developers blog concerning latency with ts3 (currently in closed alpha): http://www.goteamspeak.com/index.php?page=blogarchive&id=2

Phew, that was a long post
/me rides away on his high horse