outgoing call quality.problems

Discussion in 'UK VOIP' started by lab, Jun 13, 2006.

  1. lab

    lab Guest

    Hi

    I appear to be having real problems with the outgoing call quality of
    different VOIP providers (so im guessing its my fault) to PSTN numbers
    (havent tried SIP-SIP, except for echo tests my voipfone), my config is as
    follows

    NTL 10Mbit (i think its 600/700kilobit upload)
    Linksys PAP2
    Codecs tried gu711a/u and g729
    Asterisk 2.8 on a dedicated linux machine (although i have bypassed asterisk
    and its still the same)

    Ive tried the voipfone echo tests, and it appears to be ok, although its
    hard to tell as the echo is sent back almost immediately (i like the skype
    way better where you can talk for 10 seconds then have it echo'd back),
    however when i phone a PSTN (usually a 01252 number), the caller always says
    that the voice is breaking up etc. and to phone them back using a PSTN
    phone. This seems to happen on 3 of the VOIP providers i have tried so far
    (SIPGATE/VOIPFONE/SIPPHONE), however on all these calls, the incomming call
    quality (the voice i hear) is always perfect.

    Is there anybody who can help me and tell me what to look for to perhaps fix
    the issue. I also sent a message to voipfone support to see if they can
    help. Also during all phone calls the internet line is not being used at
    all.

    Many thanx for any answers.
    lab, Jun 13, 2006
    #1
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  2. lab

    Jono Guest

    lab presented the following explanation :
    > Hi
    >
    > I appear to be having real problems with the outgoing call quality of
    > different VOIP providers (so im guessing its my fault) to PSTN numbers
    > (havent tried SIP-SIP, except for echo tests my voipfone), my config is as
    > follows
    >
    > NTL 10Mbit (i think its 600/700kilobit upload)
    > Linksys PAP2
    > Codecs tried gu711a/u and g729
    > Asterisk 2.8 on a dedicated linux machine (although i have bypassed asterisk
    > and its still the same)
    >
    > Ive tried the voipfone echo tests, and it appears to be ok, although its hard
    > to tell as the echo is sent back almost immediately (i like the skype way
    > better where you can talk for 10 seconds then have it echo'd back), however
    > when i phone a PSTN (usually a 01252 number), the caller always says that the
    > voice is breaking up etc. and to phone them back using a PSTN phone. This
    > seems to happen on 3 of the VOIP providers i have tried so far
    > (SIPGATE/VOIPFONE/SIPPHONE), however on all these calls, the incomming call
    > quality (the voice i hear) is always perfect.
    >
    > Is there anybody who can help me and tell me what to look for to perhaps fix
    > the issue. I also sent a message to voipfone support to see if they can
    > help. Also during all phone calls the internet line is not being used at
    > all.
    >
    > Many thanx for any answers.


    What router are you using?

    What ports (if any) have you forwarded to asterisk?
    Jono, Jun 13, 2006
    #2
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  3. lab

    lab Guest

    > What router are you using?
    >
    > What ports (if any) have you forwarded to asterisk?
    >
    >


    Its a linksys WRT54GX with firmware 1.02.10 (latest i think).
    I've forwarded ports 10k-11k for asterisk (and set asterisk to use only
    these ports in the udp config)
    If im using the PAP2 directly i forward ports 16384-16482 (there the RTP
    min/max in the pap2 config)

    The port forwarding must be working correctly in all cases, as the voice i
    hear (which is the forwarded ports) is always crystal clear.
    lab, Jun 13, 2006
    #3
  4. lab

    Jono Guest

    lab formulated on Tuesday :
    >> What router are you using?
    >>
    >> What ports (if any) have you forwarded to asterisk?
    >>
    >>

    >
    > Its a linksys WRT54GX with firmware 1.02.10 (latest i think).
    > I've forwarded ports 10k-11k for asterisk (and set asterisk to use only these
    > ports in the udp config)
    > If im using the PAP2 directly i forward ports 16384-16482 (there the RTP
    > min/max in the pap2 config)
    >
    > The port forwarding must be working correctly in all cases, as the voice i
    > hear (which is the forwarded ports) is always crystal clear.


    I have yet to come accross a 1-way (or similar) audio problem which is
    not, in some way related to port forwarding or firewall.

    Port 5060 needs forwarding.

    Have you looked into the third party firmware for your router?

    There is a voip version of DD-WRT, which has a simple SIP server built
    in. No port forwarding (for sip) is required.
    Jono, Jun 13, 2006
    #4
  5. lab

    lab Guest

    > I have yet to come accross a 1-way (or similar) audio problem which is
    > not, in some way related to port forwarding or firewall.
    >
    > Port 5060 needs forwarding.
    >
    > Have you looked into the third party firmware for your router?
    >
    > There is a voip version of DD-WRT, which has a simple SIP server built in.
    > No port forwarding (for sip) is required.


    Im going to try it again tommorow without the router plugged in, see where i
    get. I forgot to mention that i did have 5060 forwarded before. If it is
    a router issue, then could somebody provide me with the name of a working
    good router :), doesnt need anything fancy as im probably going to keep the
    WRT54GX for the great wireless coverage its giving me at the moment.
    lab, Jun 13, 2006
    #5
  6. lab

    Jono Guest

    It happens that lab formulated :
    >> I have yet to come accross a 1-way (or similar) audio problem which is not,
    >> in some way related to port forwarding or firewall.
    >>
    >> Port 5060 needs forwarding.
    >>
    >> Have you looked into the third party firmware for your router?
    >>
    >> There is a voip version of DD-WRT, which has a simple SIP server built in.
    >> No port forwarding (for sip) is required.

    >
    > Im going to try it again tommorow without the router plugged in, see where i
    > get. I forgot to mention that i did have 5060 forwarded before. If it is a
    > router issue, then could somebody provide me with the name of a working good
    > router :), doesnt need anything fancy as im probably going to keep the
    > WRT54GX for the great wireless coverage its giving me at the moment.


    You /may/ already have a great router - check out loading 3rd party
    firmwear & gain even greater control over the wireless side.......and
    get a sip server to boot.
    <http://www.dd-wrt.com/dd-wrtv2/index.php?link=downloads>

    Perhaps someone else can confirm whether the WRT54GX is upgradeable, or
    is it one that has been crippled with the reduced onboard memory?
    Jono, Jun 13, 2006
    #6
  7. lab

    Mark Guest

    On Tue, 13 Jun 2006 20:34:20 +0100, "lab" <> wrote:

    >> I have yet to come accross a 1-way (or similar) audio problem which is
    >> not, in some way related to port forwarding or firewall.
    >>
    >> Port 5060 needs forwarding.
    >>
    >> Have you looked into the third party firmware for your router?
    >>
    >> There is a voip version of DD-WRT, which has a simple SIP server built in.
    >> No port forwarding (for sip) is required.

    >
    >Im going to try it again tommorow without the router plugged in, see where i
    >get. I forgot to mention that i did have 5060 forwarded before. If it is
    >a router issue, then could somebody provide me with the name of a working
    >good router :), doesnt need anything fancy as im probably going to keep the
    >WRT54GX for the great wireless coverage its giving me at the moment.


    I have exactly the same problem, also on NTL and also on Voipfone.
    Try rebooting your CM to see if that fixes or helps it, even
    temporarily.

    My problem is intermittent and is a real pain as I've had the guys at
    Voipfone delve pretty deeply into this. One possible thing to try
    (Voipfone suggestion) is avoiding any DECT phones attached to ATAs -
    which some evidence suggests might be a source of problems.

    My theory is that my cable modem (an ancient Terayon TJ210) is the
    problem. I am going to take my ATA to the in-laws one day to run it
    on a DSL connection.
    Mark, Jun 13, 2006
    #7
  8. lab

    lab Guest

    > My theory is that my cable modem (an ancient Terayon TJ210) is the
    > problem. I am going to take my ATA to the in-laws one day to run it
    > on a DSL connection.


    Mark, just to let you know, ive tried with the old surfboard 3100 modems
    (5-6 years old) and a new NTL 250 Home modem, both gave the same results.
    Maybe its NTL :(
    lab, Jun 13, 2006
    #8
  9. lab

    Ivor Jones Guest

    "Mark" <> wrote in message
    news:eek:

    [snip]

    > My problem is intermittent and is a real pain as I've had
    > the guys at Voipfone delve pretty deeply into this. One
    > possible thing to try (Voipfone suggestion) is avoiding
    > any DECT phones attached to ATAs - which some evidence
    > suggests might be a source of problems.


    Cobblers. I've been using DECT phones for over 18 months on a variety of
    ATA's (and providers, although not Voipfone) with no problems. Sounds like
    a fob-off to me.

    Ivor
    Ivor Jones, Jun 14, 2006
    #9
  10. lab

    news Guest

    In message <448ed432$0$13951$>, lab
    <> writes

    >NTL 10Mbit (i think its 600/700kilobit upload)


    Just a hunch. I wonder if it is to do with jitter on the uplink. See the
    ADSL interleaving thread currently elsewhere in this group. (I know NTL
    doesn't use ADSL, but that thread may trigger a thought).

    --
    Ian
    news, Jun 16, 2006
    #10
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