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Home > Archive > Voice Over IP in UK > June 2006 > outgoing call quality.problems
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outgoing call quality.problems
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| 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.
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| 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?
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| > 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.
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| lab formulated on Tuesday :
>
> 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.
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| > 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.
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| It happens that lab formulated :
>
> 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?
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| On Tue, 13 Jun 2006 20:34:20 +0100, "lab" <billy@microsoft.com> wrote:
>
>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.
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| > 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 
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| Ivor Jones 2006-06-14, 1:11 am |
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"Mark" <markincambs@yahoo.co.uk> wrote in message
news:o29u82h5f977u98fkni1lk6vvqkfo2oqal@
4ax.com
[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
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| In message <448ed432$0$13951$9a6e19ea@unlimited.newshosting.com>, lab
<billy@microsoft.com> 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
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