Latency expectations through 3CX server

Discussion in '3CX Phone System - General' started by fjkeith, Nov 8, 2007.

  1. fjkeith

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    I have been working with nexVortex (my test VOIP provider) to address some significant latencies in my calls:

    IP Phone-to-IP phone (internal, LAN only) - about 200 mS acoustic delay using Polycom 600 phones running G711u

    IP Phone to external PSTN phone via nexVortex SIP trunk - about 350 mS acoustic delay

    nV tech support advised that my internal delay should be on the order of 1mS on the LAN. I will setup to sniff tomorrow, but in the meantime I installed Wireshark 0.99.6 on my 3CX server directly, and it looks like I'm seeing a delay in/out of the server itself of about 130 mS:

    Analysis of a 3-4 sec call between two extensions (Polycom600/G711u):

    I scrolled through the packets from phone #1 (10.30.8.110) to the server at 10.30.8.10 until I found one where the payload was not all FF FF FF FF (which I assume is silence). The destination phone is at 10.30.8.108.

    Here's what I found:

    Packet 31, 12.092353 sec, 10.30.8.110 to 10.30.8.10, payload =
    0000 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ................
    0010 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ................
    0020 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ................
    0030 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ................
    0040 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ................
    0050 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ................
    0060 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ................
    0070 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ................
    0080 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ................
    0090 ff ff ff ff ff ff 7e 7e 7e 7e 7d 7d 7c 7d 7e ff ......~~~~}}|}~.

    Packet 34, 12.112256 sec, 10.30.8.110 to 10.30.8.10, payload =
    0000 fe fd fd fd fc fd fd fd fd fd fe 7e 7d 7d 7d 7d ...........~}}}}
    0010 7e 7e fe 7e 7d 7b 79 7a 7a 7a 7a 7a 7a 7c 7c 7c ~~.~}{yzzzzzz|||
    0020 7d 7e ff fd fd f9 f8 f9 f8 f9 f9 f9 f9 f9 f9 fa }~..............
    0030 fc fb fc fd fd 7d 7b 7b 7d 7d 7d fe fe fe 7e 7c .....}{{}}}...~|
    0040 7b 7c 7d 7e ff 7d 7c 7b 7b 7a 79 7a 79 7a 7d 7e {|}~.}|{{zyzyz}~
    0050 7e ff 7e ff 7d 7d ff fe fd fd fb fd fd fc fb fb ~.~.}}..........
    0060 fc fd fc fb fd fd fd 7e 7e 7d 7e 7d 7d 7e 7c 7b .......~~}~}}~|{
    0070 7a 79 76 75 76 78 7a 7d 7d 7e fe 7e fe 7e fd fb zyvuvxz}}~.~.~..
    0080 fb fa fb f8 f7 f7 f7 f6 f7 f9 f8 f8 f9 f8 f9 fa ................
    0090 fc fd ff 7c 7d 7a 78 77 76 76 73 74 74 74 77 78 ...|}zxwvvstttwx

    Packet 53, 12.221557 sec, 10.30.8.10 to 10.30.8.108, payload =
    0000 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ................
    0010 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ................
    0020 ff ff ff ff ff ff 7e 7e 7e 7e 7d 7d 7c 7d 7e ff ......~~~~}}|}~.
    0030 fe fd fd fd fc fd fd fd fd fd fe 7e 7d 7d 7d 7d ...........~}}}}
    0040 7e 7e fe 7e 7d 7b 79 7a 7a 7a 7a 7a 7a 7c 7c 7c ~~.~}{yzzzzzz|||
    0050 7d 7e ff fd fd f9 f8 f9 f8 f9 f9 f9 f9 f9 f9 fa }~..............
    0060 fc fb fc fd fd 7d 7b 7b 7d 7d 7d fe fe fe 7e 7c .....}{{}}}...~|
    0070 7b 7c 7d 7e ff 7d 7c 7b 7b 7a 79 7a 79 7a 7d 7e {|}~.}|{{zyzyz}~
    0080 7e ff 7e ff 7d 7d ff fe fd fd fb fd fd fc fb fb ~.~.}}..........
    0090 fc fd fc fb fd fd fd 7e 7e 7d 7e 7d 7d 7e 7c 7b .......~~}~}}~|{

    Thus, from packet 31 to packet 53 the time delay was 12.221 - 12.092 = 0.129 sec, or 129 mS just to get the data in and out of the server. This does not account for any LAN delays, or processing/queueing/conversion delays in the phones.

    My roughly measured ACOUSTIC delay (phone to phone) is about 200 mS.

    What might be causing this delay in the 3CX server? Running Version 3.1.2434

    thanks!
    Joe
     
  2. fjkeith

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    Followup question:

    What, if any, codec processing does the 3CX server do on packets passing between two G711u phones? Are the data streams simply passed through, or are they in any way "re-converted" that might add additional delay?

    Are there any particular setup notes re. tweaking the Windows OS and/or the PC platform to achieve minimum delay?

    thanks!
     
  3. fjkeith

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    More test data:

    I uninstalled the packet capture and remote desktop apps from my 3CX server as a "just in case" measure. Setup a test network limited to two phones (Polycom600, G711u, 20mS payload), the 3CX PC, and my laptop as packet sniffer.

    Total in-to-out delay just through the 3CX server = 130 mS. Acoustic delay still about 200 mS

    When I uncheck the "bind to media server" box for the two phones in the 3CX setup, the packet sniffer shows that the RTP packets are direct phone-to-phone (they do not route through the 3CX server). Delay is so short I cannot reliably measure it, definitely less than 100 mS acoustic delay. With 20 mS payloads, my one-way delay minimum is 40 mS, plus any phone processing time and LAN delays.

    What are the implications of "unbinding" from the media server? I had tested the oreka phone recording software, which worked great, but with no packets through the server that would be a bust. Will call logs still be as complete as if the extensions were bound to the media server?

    I made an external call via the nV SIP trunk connection with the extensions unbound. Got one-way outbound audio. Not sure about the port-mapping implications of having the return audio channels attempting to go to various phones on our LAN.

    We have tried to setup a V4 server on a separate box, but we are having problems getting the phones to register.

    Please advise. thanks!
     
  4. fjkeith

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    BTW, the packets in/out of 3CX are NOT identical, which means that the 3CX software is de-converting and re-converting the audio stream. A better solution might be to pass them through identically, but "sniff" a copy to support recording, etc.
     
  5. scooby

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    Hi I am suffering high latency and wondered if you had made any further progress on your investiagtions. I am consdering trying a different hardware config to see if that helps, I have checked the various drivers and updated the NIC, but is still very high when say coming in via the pstn gatway (soundstream) to an extension, at first i thought it was down to the voip provider i was using, but now i ahve added the pstn gatway I see no imprvement,

    cheers

    Andrew
     
  6. fjkeith

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    For all gateways and VOIP providers, uncheck the "PBX delivers audio" option for lowest latency. Let me know if that helps.
    thanks, Joe
     

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