Why large delay takes place if using "Transcoding mode"?

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Alexord

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Hi,

I have 3cx installed on separate server, I can see that neither CPU nor memory usage is high but I still have delay in conversation independently of codec. Does it point to network problem or something wrong with my server? Any ideas?

In proxy mode I still have delay but less noticeable (100-150 ms in proxy mode contra 500 - 700 ms in Transcoding mode). All SIP test shows good results - approximately 65 ms delay in trace route and 0% packet loss.
 
What type of delay are you talking about for example is it:

h--e--l--l--o--

or is it

hello-----how----are----your


Please advise.
H.
 
which are the characteristics of the system that you use?
 
If I call myself on cell phone and say hello, I'll hear this "hello" on other phone after some time. But quality of sound is good.

I use Pentium 4 - 2400 Mhz, 1 Gb RAM, Windows Server 2003 R2 behind IPtables firewall. This machine running 3CX only.
 
What sort of delay are we talking about??? 5 secs, 1 sec, .5 sec??? You should expect a short (.5 sec) or so especially on a cell. If it's more it could be bandwidth. Whats your net speed and was anything running in the background that would be consuming your bandwidth?
 
Yes, I'm talking about delay proximately 0.5 sec.
This delay appears only in "transconing mode" and even in local calls. If PBX deliver audio is off, there no delay at all in local calls and a little in external. So I don't think that is bandwidth or firewall problem due to same sort of thing in local network...
What could cause this? Slow machine? If I upgrade it to more faster one, does it helps? There is no anti-virus or any other software except 3CX on this server.
 
I don't think that depend on the characteristics of the PC. this is good
 
This delay arises because media server adaptively sets up jitter buffer. If both streams are even and don't deviate from timer - jitter will be less. If streams are not regular (time between packets is not stable) - jitter will increase. Otherwise sound quality will degrade noticeable.
 
It sounds reasonable, but why its happened even in local network? And also my internet connections tests always show 3-4 ms average jitter...
 
It depends on the quality of the server, internet connection, sip phones, voip provider being used. Lots of variables that come into play here. On the internal network, it should not operate in transcoding mode, not if you are using quality phones and have not changed the settings.
 
Is there a possibility to turn off the jitter buffer ?

I think a "PBX" beeing "in the middle" beteween to partys shouldn't do jitter buffering, unless specificaly ask to do so.

The best place for jitter buffering is on the devices.
 
When in proxy mode Media server doesn't add any jitter. In case of transcoding jitter is inevitable.
 
Sure, transcoding does add some jitter.

But I see no good reason to perform full blown jitter buffering except for devices that have have no or no good jitter buffer, or if 3CX itself is the endpoint.

It just adds delay with no real benefit.
If packets arrive untimely at 3CX and 3CX is emitting them untimely subsequently the jitter buffer of the endpoint can handle that problem.
 
As I already told 3CX doesn't have jitter in proxy connections.
When transcoding - jitter is inevitable, because MS needs some ready protion of decoded data before it can encode them and send out. So, if one endpoint sends 20ms packets, and another endpoint waits for 30ms packets, and second packet arrives in 40ms after first one - what MS should do? Send silence or what?
Please, let's not discuss technical questions. Believe me, we thought a lot before implement it. If you're good specialist in that area - make a suggestion and we will implement it if find it reasonable. All other talks are in vain, I suppose.
 
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