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uk.telecom.voip (UK VOIP) (uk.telecom.voip) Discussion of topics relevant to packet based voice technologies including Voice over IP (VoIP), Fax over IP (FoIP), Voice over Frame Relay (VoFR), Voice over Broadband (VoB) and Voice on the Net (VoN) as well as service providers, hardware and software for use with these technologies. Advertising is not allowed.

outgoing call quality.problems



 
 
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  #1  
Old June 13th 06, 04:05 PM posted to uk.telecom.voip
lab
external usenet poster
 
Posts: 10
Default outgoing call quality.problems

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.


  #2  
Old June 13th 06, 06:46 PM posted to uk.telecom.voip
Jono
external usenet poster
 
Posts: 151
Default outgoing call quality.problems

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?


  #3  
Old June 13th 06, 07:18 PM posted to uk.telecom.voip
lab
external usenet poster
 
Posts: 10
Default outgoing call quality.problems

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.


  #4  
Old June 13th 06, 08:04 PM posted to uk.telecom.voip
Jono
external usenet poster
 
Posts: 151
Default outgoing call quality.problems

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.


  #5  
Old June 13th 06, 08:34 PM posted to uk.telecom.voip
lab
external usenet poster
 
Posts: 10
Default outgoing call quality.problems

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.


  #6  
Old June 13th 06, 09:31 PM posted to uk.telecom.voip
Jono
external usenet poster
 
Posts: 151
Default outgoing call quality.problems

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?


  #7  
Old June 13th 06, 10:44 PM posted to uk.telecom.voip
lab
external usenet poster
 
Posts: 10
Default outgoing call quality.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, 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


  #8  
Old June 14th 06, 01:32 AM posted to uk.telecom.voip
Ivor Jones
external usenet poster
 
Posts: 3,969
Default outgoing call quality.problems


"Mark" wrote in message
news
[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


  #9  
Old June 16th 06, 03:06 PM posted to uk.telecom.voip
news
external usenet poster
 
Posts: 74
Default outgoing call quality.problems

In message om, 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
 




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