Very poor softphone voice quality even on LAN

Discussion in 'VOIP' started by Ed M, Nov 25, 2004.

  1. Ed M

    Ed M Guest

    I'm experimenting with using SIP and softphones on our LAN (and eventually
    WAN.)
    I'm amazed at how variable the quality is even on a lightly-loaded LAN.
    I have two test Windows 2000 machines connected to the same HP Procurve
    4104GL switch (which is supposed to have QOS support enabled by default) at
    100mbit with a softphone loaded on each.
    Conversation is great until one of the machines does almost any kind of
    network access. Then there is immediate "crunchiness" or skipping of audio.
    I've tried a variety of softphones (eyeP media, SoftJoy SJPhone) and codecs
    (G.729, G711ulaw) but all seem to have this problem.
    Am I missing something with configuring QOS with Windows 2000 Professional
    itself? I tried enabled 802.1p support on the NIC's but this didn't help
    either.
    Ed M, Nov 25, 2004
    #1
    1. Advertising

  2. Ed M

    Mitel Lurker Guest

    802.1p/q has to be enabled throughout the LAN, not just on the NICs. Until
    you are able to prioritize your VOICE packets, it's going to sound like
    crap when competing for bandwidth with your data. VOIP either needs to be
    deployed ALONE on it's own separate, isolated LAN (or VLAN) or else
    requires priority tagging. Until you do that, you're not going to be happy
    with VOIP. Anytime VOIP has to contend for bandwidth with data
    transmission, the voice quality will suffer because it cannot deal with
    lost packets or late packets.

    In article <Exspd.5242$> "Ed M"
    <> writes:

    >I'm experimenting with using SIP and softphones on our LAN (and eventually
    >WAN.)
    >I'm amazed at how variable the quality is even on a lightly-loaded LAN.
    >I have two test Windows 2000 machines connected to the same HP Procurve
    >4104GL switch (which is supposed to have QOS support enabled by default) at
    >100mbit with a softphone loaded on each.
    >Conversation is great until one of the machines does almost any kind of
    >network access. Then there is immediate "crunchiness" or skipping of audio.
    >I've tried a variety of softphones (eyeP media, SoftJoy SJPhone) and codecs
    >(G.729, G711ulaw) but all seem to have this problem.
    >Am I missing something with configuring QOS with Windows 2000 Professional
    >itself? I tried enabled 802.1p support on the NIC's but this didn't help
    >either.
    >
    >
    Mitel Lurker, Nov 26, 2004
    #2
    1. Advertising

  3. "Ed M" <> wrote in
    news:Exspd.5242$:

    > Conversation is great until one of the machines does almost any kind
    > of network access.


    You must have some problem on your network. If a single VoIP call (in any
    codec) doesn't have acceptable quality on such a simple setup, there's got
    to be a problem. Especially if the problems are comparable between G.711
    and G.729. What does the CPU load on the PC's show?

    --
    Andreas
    Andreas Sikkema, Nov 26, 2004
    #3
  4. Ed M

    DevilsPGD Guest

    In message <> Mitel Lurker
    <wdg@[206.180.145.133] wrote:

    >802.1p/q has to be enabled throughout the LAN, not just on the NICs. Until
    >you are able to prioritize your VOICE packets, it's going to sound like
    >crap when competing for bandwidth with your data. VOIP either needs to be
    >deployed ALONE on it's own separate, isolated LAN (or VLAN) or else
    >requires priority tagging. Until you do that, you're not going to be happy
    >with VOIP. Anytime VOIP has to contend for bandwidth with data
    >transmission, the voice quality will suffer because it cannot deal with
    >lost packets or late packets.


    Sounds like you need to upgrade your switches (or stop using hubs?).

    I some VoIP gear (3 Vonage lines, another hardware SIP device,
    softphones on a few machines, plus some audio streaming) across a 100Mb
    network.

    All of the VoIP gear (Vonage and internal VoIP alike) are all connected
    to my primary network and there is no internal packet prioritizing
    taking place.

    I regularly transfer data between servers and between workstations and
    two servers, bottlenecking at the 100Mb switch port connecting the
    server (And yes, a gigE upgrade is being budgeted, although only for the
    servers initially and one or two workstations -- The servers mirror
    large chunks of data between them though, so they'll benefit from the
    gigE even if the uplink to the network has to stay at 100Mb for the
    moment.

    I've pushed 200-300Mb's through the switch (various machines
    transferring to other machines), plus saturated my upstream connection
    (to approximately 90% of it's capacity, which is 100% of my rate
    limiting -- The line is configured for 5Mb/1Mb), then picked up a Vonage
    line and made a crystal clear phone call.

    The ONLY packet prioritization takes place on the border router /
    firewall at this time, the only thing I've done on the switch is to
    prioritize traffic from the firewall to the rest of the LAN. Since the
    WAN side of the firewall will only ever have up to 5Mb inbound, there is
    no risk of this traffic saturating internally.

    On the firewall I prioritize ACKs, VoIP traffic, ICMP, DNS queries,
    HTTP+SMTP+IMAP user access, then all other traffic (in that order).

    ICMP is only prioritized for testing purposes -- I can maintain 20ms
    ping times to my provider's first hop regardless of internal traffic,
    and my provider has sufficient bandwidth that nothing on their network
    is bottlenecking, so there is no need for QoS (As much as it would be
    nice, it's not affordable at this time)

    All that being said, before I upgraded to a "real" switch I was using
    some cheapo SOHO Linksys gear and definitely noticed a performance hit
    on the Vonage equipment if my servers were performing nightly backups
    (to each other)

    --
    A gun isn't a weapon; it's a tool. Like a harpoon, or a hammer or an
    alligator. You just need more education on this subject.
    -- Homer Simpson
    DevilsPGD, Nov 26, 2004
    #4
  5. Ed M

    Ed M Guest

    Thanks for all the replies to my question
    I'll answer the various questions and then see if anyone has any more ideas
    for me.

    >>>it's going to sound like crap when competing for bandwidth with your

    data.
    Yes but these are 100 mbit connections. I can see this happening on a WAN
    but on 100mbit LAN connections I would think that there would be plenty of
    bandwidth to spare.

    >>>Sounds like you need to upgrade your switches (or stop using hubs?).

    I'm using HP Procurve switches. According to various reviews (Tolly Group
    etc.) these are as good as Catalysts for VOIP QOS purposes.

    >>>What does the CPU load on the PC's show?

    I'm beginning to wonder whether softphones can really deliver "toll quality"
    voice. Its seems that almost ANY activity on the PC's (from what I can tell
    even ones not involving network access) causes some audio degradation. I'm
    testing by making a call using the G711 codec (I've tried both U and A law)
    and putting myself on hold and listening to music. This codec is
    indistinguishable in quality from a POTS call (vs. G.729 which is nowhere
    near toll quality) but then there are the drop-outs. My test PC's are
    fairly high end (1+ ghz CPU's with plenty of RAM)

    I then I wonder whether other customers of the softphones I've been trying
    expect toll quality voice. Is it possible that some people's expectations
    have gone down in this age of cell phones with their highly compressed audio
    and free VOIP where people are happy to have free calling even if it isn't
    perfect? The application I'm working on is for a sales department where the
    audio quality needs to be as close to toll quality as possible.
    Ed M, Nov 26, 2004
    #5
  6. Ed M

    Mitel Lurker Guest

    In article <Q_Fpd.5282$> "Ed M"
    <> writes:


    >I'm beginning to wonder whether softphones can really deliver "toll quality"
    >voice. Its seems that almost ANY activity on the PC's (from what I can tell
    >even ones not involving network access) causes some audio degradation.


    That in itself could be a clue.
    Check the NIC card configuration on the PC to make absolutely sure you are
    configured (locked) for 100 megs and full duplex, not 10/100 and not half
    duplex and not auto-negotiate. Also make sure that nothing is connected to
    a HUB. Any HUB + VOIP = Misery.

    What else is the PC doing? Can you arrange to halt all other PC activity
    and run the softphone application by itself?

    >I then I wonder whether other customers of the softphones I've been trying
    >expect toll quality voice. Is it possible that some people's expectations
    >have gone down in this age of cell phones with their highly compressed audio
    >and free VOIP where people are happy to have free calling even if it isn't
    >perfect? The application I'm working on is for a sales department where the
    >audio quality needs to be as close to toll quality as possible.


    Mitel's YA (Your Assistant) and YAPro softphones, at least in release 2.1
    do not seem to have this problem. I'm running one on my AMD 2400 at home
    over a Linksys BEFSR41 router and VPN back to the Mitel 3300 VOIP pbx at
    the office and making both local and LD calls with no observed issues and
    with what I would certainly describe as "toll quality". The lone exception
    has been when listening to MOH you can hear occasional dropouts and
    scratchiness, but live voice is fine.
    Mitel Lurker, Nov 26, 2004
    #6
  7. Ed M

    Hank Karl Guest

    On Thu, 25 Nov 2004 21:55:48 GMT, "Ed M"
    <> wrote:

    >I'm experimenting with using SIP and softphones on our LAN (and eventually
    >WAN.)
    >I'm amazed at how variable the quality is even on a lightly-loaded LAN.
    >I have two test Windows 2000 machines connected to the same HP Procurve
    >4104GL switch (which is supposed to have QOS support enabled by default) at
    >100mbit with a softphone loaded on each.
    >Conversation is great until one of the machines does almost any kind of
    >network access. Then there is immediate "crunchiness" or skipping of audio.
    >I've tried a variety of softphones (eyeP media, SoftJoy SJPhone) and codecs
    >(G.729, G711ulaw) but all seem to have this problem.
    >Am I missing something with configuring QOS with Windows 2000 Professional
    >itself? I tried enabled 802.1p support on the NIC's but this didn't help
    >either.
    >

    Sounds like you need better test tools. www.voiptroubleshooter.com
    has a list of them.

    If you're doing this on a limited budget, try using a third machine to
    do the heavy network access. This will help you determine whether its
    a network issue or an issue with your local machine (e.g. not enough
    CPU to do both VoIP and the data stuff).

    If it's a network issue, check out www.testyourvoip.com and try the
    voice quality tests there.
    Hank Karl, Nov 27, 2004
    #7
  8. Ed M

    Guest


    > I have two test Windows 2000 machines connected to the same HP Procurve
    > 4104GL switch (which is supposed to have QOS support enabled by default)


    Don't assume. Go look at the configuration and make sure. Also
    having QOS honored on the switch doesn't mean anything if the packets
    aren't tagged to begin with. Does your softphone software support QOS
    tagging or TOS tagging? The HP switch I believe will do QOS based on
    TOS tagging.


    >>>>it's going to sound like crap when competing for bandwidth with your

    >data.
    >Yes but these are 100 mbit connections. I can see this happening on a WAN
    >but on 100mbit LAN connections I would think that there would be plenty of
    >bandwidth to spare.


    It's not just overall bandwidth. It's also latency which can be
    affected by bursts of traffic, both before it hits the NIC and after
    it hits the network.
    , Nov 27, 2004
    #8
  9. Ed M

    Ed M Guest

    I think I found the problem
    The three different Dell PC's I tested (two Dimension desktops and a
    Latitude laptop) must have very low-end sound cards built into them. Maybe
    they are software based? I tested using a Dell Optiplex using the same
    softphone on the same network and the sound was perfect.

    Then I tested using a USB headset adaptor from GN Netcom (GN8110 USB) on the
    PC's that had trouble before and using that the quality is great.
    Apparently there is a DSP built into the cord that offloads the processing
    from the PC. So it looks like all along my problems were caused by the
    Softphones trying to use the low-end built-in sound cards. Strange that I
    see no other newsgroup postings about this or any warnings from the
    softphone vendors..... Everyone focuses on network QOS (which is very
    important but not the whole story.)
    Thanks for the input everyone!
    Ed M, Nov 27, 2004
    #9
    1. Advertising

Want to reply to this thread or ask your own question?

It takes just 2 minutes to sign up (and it's free!). Just click the sign up button to choose a username and then you can ask your own questions on the forum.
Similar Threads
  1. Sven Holm
    Replies:
    0
    Views:
    1,502
    Sven Holm
    Sep 7, 2003
  2. Sven Holm
    Replies:
    0
    Views:
    1,413
    Sven Holm
    Sep 9, 2003
  3. Sven Holm
    Replies:
    5
    Views:
    4,496
    Arnold Ligtvoet
    Sep 12, 2003
  4. Ramon F Herrera

    Asterisk IVR has very poor voice quality

    Ramon F Herrera, Sep 7, 2005, in forum: VOIP
    Replies:
    4
    Views:
    3,009
    sevana
    Aug 13, 2009
  5. Martin
    Replies:
    5
    Views:
    2,395
    alexd
    Feb 15, 2006
Loading...

Share This Page