Asterisk Voice Quality

Discussion in 'General' started by e.poulogiannis, Mar 13, 2009.

  1. e.poulogiannis

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    Dear all,

    I have been experiencing some problems in several of my asterisk installations (v1.2). Specifically I have some low voice quality problems. The problem does not happen on some regular basis that I can research.

    I would like to know what are those factors that could cause low sound quality. The sound problems are: sound gaps during the conversation, the remote party complains that our voice sounds very faint, the softphone user complains that the remote party sounds very faintly.

    All my users use X-Lite and the PBX is an Asterisk v1.2. The codecs that are used are only a-law.


    Thank you in advance for your help
     
  2. Patrick_elx

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    if you are using SIP trunk one of the usual suspect will be:
    upload speed / latency
    do you have QoS on your network?
    is the sound bad both way or only one way?
     
  3. e.poulogiannis

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    Hello again

    There is no sip trunking with the provider if you mean this. The connection with the provider is through an E1 Interface with a Digium card. I too thought that the problem could be with the network. I've used the QoS settings of X-Lite but i have not seen a real improvement yet. What else do you think i could do on the network configuration to help?
     
  4. ramoncio

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    You usually need to setup qos in the main router. This gives high priority to your sip traffic.
     
  5. e.poulogiannis

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    Thank you for that I will try to see what the router settings are. Just another quick question. To my understanding the router should affect internal calls as well right? Dialing from those extensions to another extension should give the same problem. Is there any chance that this is a Provider related issue as well?
     
  6. Chilling_Silence

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    Its most likely related to X-Lite itself, potentially the headset, potentially the soundcard, potentially other software on the PC affecting the Softphone.. There's *so* much that can go wrong with Softphones

    Try OpenWengo as well as ZoIPer Free and see which yields best results. I'd recommend OpenWengo personally :)
     
  7. e.poulogiannis

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    Actually i never had any problems with X-Lite. I use X-Lite on many installations (including the PBX we use in the company I work for) and I never had so many problems with that. I think that if it is workstation related, the problem is most likely on the on the hardware. I can't think of a software that could cause problems on the softphone. At least not problems that arise one time and not the other.

    In this installation, they use Vnc monitoring of the users desktop but I think that if this affected the voice quality, it would have happened on all workstations, all the time.

    Using wireshark i have seen that the network layout creates a lot of latency and jitter. Currently the don't have a router that would shape the voip traffic. I'm more inclined that the problem here lies completely on the network. I will keep this thread updated on my findings on this subject but as i said I'm more inclined that the problem lies on the network

    Thanks for your suggestions on the softphones. I've used Zoiper in the past, never really liked it. I'll check out QuteCom (formerly known as WengoPhone :p) and see :)
     
  8. Bob

    Bob

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    There are many, many factors that will effect Voice Quality, including the architecture of the Network.

    Small flat networks (e.g. Single Network switch)
    ===================
    Look at Switches e.g. overload - cheap switches may not have a back plane that can support all lines operating at 100Mb and pushing heavy loads (e.g. cad drawings, etc)
    Disable jumbo packet handling (normally on higher end switches), as other devices may not handle it
    Perform a network test on each machine to locate serious differences in transmission time.
    make sure that you have the latest Network drivers on each machine

    Larger multi-switch networks
    ============================
    Uplinks between switches (are these being overloaded)
    Has someone linked two ports on one switch to two ports on another (thinking that they were aggregating them and the switches didn't support it.) DONT laugh it has happened!!
    Again jumbo packets disable

    One of the most common issues is that many people believe that they have no network issues as everything works ok. With TCP based communications, which generally covers most internet applications, such as Internet Browsing, mail etc, you never get to see the retries on a badly constructed network. With UDP (most voice and video apps) it shows up very well. Some applications hide it such as Video, as applications buffer the stream, and basically this compensates for a poorly operating network.

    As a guide, a client with a switch with two ports linked to another, with a switch that could not aggregate the ports ran like this for 7-8 months, believing their network to be running well. At certain times their voice quality suffered, and it wasnt until it was narrowed down that the VoIP connections on one switch were fine (also the switch with the Asterisk system), and the other suffered occasional loss of quality.

    Anyhow just passing some thoughts...hope it helps...

    Bob
     
  9. e.poulogiannis

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    Thank you everyone for your responses.

    For the past days i have been doing some tests to check on the network quality. My guess is that it's mostly network related and that in some workstations a change in hardware/drivers is in order. The only tools in my disposal have been is tcpdump on the server side and wireshark on the client side. Checking the pcap files with wireshark i've noticed that on the occasions where the user complained about the voice quality, the jitter was larger than 10ms.

    What seems a little strange to me is that when the network technician enabled QoS on the router, I didn't see a noticable change.

    I would just like to throw another question here as well. Could the fact that MTU is configured at 1500 on asterisk server and is 1472 on the network have anything to do with it?

    Thank you all for your much needed help.
     
  10. Chilling_Silence

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    Yup, thats happened, but mostly a format of the PC and a clean installation fixes things, get the latest drivers when you do-so and dont leave resource-hungry AntiVirus software running

    Is that just from the PC to the local asterisk server on the network? Thats horrible :( there's the problem right there ;)

    Try just pinging for 2-3 hours from each PC (Outputting to a file) and see what the jitter is:
    ping 10.1.1.123 -w 100 -t > c:\pinglog.txt

    because that will only probably affect the relationship between your Asterisk box and your ITSP, not your Asterisk box and other local clients

    Its less than ideal, but not a deal-breaker, it shouldnt be *that* bad :-/
     
  11. e.poulogiannis

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    Chilling_Silence wrote:

    There is no voip provider if that's what you mean. What the network technician was trying to do is enable qos on the local network.

    My thoughts exactly. :)
     
  12. Chilling_Silence

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    That would need to be done at the Switch level, however unless its quite congested, its rarely the issue.
     
  13. e.poulogiannis

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    Maybe I misguided you. The QoS was configured on the switch not the router. I'm sorry for the missinformation :)
     
  14. sevana

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    Hello,

    Concerning measuring Asteriks voice
    quality for various terminations, we have developed a software that is
    successfully used by some of our customers who have Asteriks installed and
    perform voice quality monitoring quite easily on their own hardware in
    this manner:

    1. Originate a call on the monitoring server, using Asterisk manager
    interface, to a server which is running an echo application
    2. Monitor both inbound and outbound legs of the call, and save as wav
    files.
    3. Use AQuA (our product) to compare the wav files.

    We would be happy to hear from those who are interested in such solution and will
    be looking forward to your feedback. Thank you!

    Best regards,
    Endre
    Sevana Oy
    http://www.sevana.fi
     

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