[asterisk-users] Outgoing PSTN calls , unusable voice quality

Mon Dec 3 11:46:03 CST 2007

Well do you have a packet filter between the asterisk box and your phone? Is
the phone or the asterisk behind a NAT? Do you have an asymmetric route for
traffic in your network? Does the media take the same path inbound and
outbound between the asterisk and the phone?  If you take a packet capture
of the voip segment of both an inbound and outbound call, how do they
differ? Are those calls using different codecs? 

If you want to rule out the analog card, construct a loop. When an inbound
call comes in, send it back out to the PSTN. Then create the call and get a
MOS score if you can.

You should also consider terminating an inbound and an outbound call against
the echo application in asterisk and see if you can further isolate the
problem to either the PSTN channel or the voip channel.

You really need to try to isolate the problem to a particular call segment.
Once you do that, make sure your test is repeatable. I would be very
hesitant to declare a hardware issue with the Sangoma card. Unidirectional
audio problems are usually caused by a Voip leg since media is handled
separately during that segment of the call. I would be extremely surprised
if this was a hardware issue with a sangoma card.

Good luck, SG 

--------------------------------------------------
Salvatore Giudice
Salvatore.Giudice at VoIPSecurityTraining.com

VoIP Security Training, LLC
http://VoIPSecurityTraining.com

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Veselin
Kantsev
Sent: Sunday, December 02, 2007 9:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Outgoing PSTN calls , unusable voice quality

Dear Salvatore and Joanna,
Thank you much for both your detailed explanations.
I will surely check my firewall configuration and logs to make sure the VoIP
traffic is passing correctly.

However, I'm a bit confused as the problems that I'm experiencing are with
calls made via the
Sangoma analog card. 
So the voice goes from the SIP phone through asterisk through the sangoma
card then 
directly into the PSTN and vice versa. There are no firewalls in the way. 

Furthermore incoming calls are OK, the problem is only with outgoing calls
when I can hear the other party well 
but they barely understand me.

Is there any major difference in the way that Incoming/Outgoing calls are
processed in the above scenario?
Any way that I could trace those processes for faults?

Thank you again.

Regards,
Veselin

Salvatore Giudice wrote: 

When you take your packet capture, you'll need to look at the sip messages
with SDP attached to get the ip's and ports used for both media streams.
Make sure that the ips are correct and that the port used can traverse
between those ip's without being blocked by a packet filter or firewall. A
lot of times, administrators will set a range of UDP ports that are allowed
to pass their packet filter for media and your pbx or phones may be using a
different range. This can cause audio loss. You'll need to eliminate that
possibility. Sometimes checking your firewall/packet filters for blocks may
also prove helpful in identifying problems. You should be aware that the
logs from certain firewall products may not be comprehensive. For example,
in the past I have seen packets dropped going through netscreens because of
invalid headers and no entries appeared in the logs. If you ultimately
believe a firewall may be blocking your traffic make sure you setup a
capture port or a span on each side of the device and verify the traffic
going to and leaving from the firewall using ethereal on a laptop or maybe a
Nixon box if you are in a large distributed environment. Never trust a
potentially broken device to report accurate information about it's
function.

TDM = Time Division Multiplexing

TDM describes how channels are separated on T1's, etc. It's common to refer
to those types of connections as TDM.
http://en.wikipedia.org/wiki/Time-division_multiplexing

--------------------------------------------------
Salvatore Giudice
Salvatore.Giudice at VoIPSecurityTraining.com

VoIP Security Training, LLC
http://VoIPSecurityTraining.com

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Veselin
Kantsev
Sent: Friday, November 30, 2007 8:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Outgoing PSTN calls , unusable voice quality

Thank you much for the prompt reply Salvatore.

Would you have the time to explain further how should I go for verifying
that SDP and RTP are OK.
Also what is reffered to as the TDM site.

Veselin

On Fri, Nov 30, 2007 at 05:01:17PM -0500, Salvatore Giudice wrote:

Take a packet capture of your VoIP segment and verify that the SDP is
correct and that the RTP is making it to the correct places. If all that
looks good and this is a straight out quality problem, then you need to
figure out if it's happening on the voip side or on the TDM side. You

should

make calls (with captures) VoIP to Voip passing the media through your
asterisk and also try routing a tdm call in and back out. If you have the
equipment, take a mos score of the TDM loop.

Without any of the above, you will not be able to isolate the issue.

--------------------------------------------------
Salvatore Giudice
Salvatore.Giudice at VoIPSecurityTraining.com

VoIP Security Training, LLC
http://VoIPSecurityTraining.com

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Veselin
Kantsev
Sent: Friday, November 30, 2007 2:47 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Outgoing PSTN calls , unusable voice quality

Hello,
I have an Asterisk running with a Sangoma A200 card with Hardware Echo 
cancelling connected to the UK PSTN.
If a PSTN call comes in, voice both ways is OK, however if an outgoing 
call over the PSTN is made I can hear the other party OK but they can 
not, they can barely understand what I am saying, my voice is unclear 
fading and skipping.
Internal SIP and IAX2 calls are OK, incoming/outgoing calls over IAX2 
are OK too. I've tried gsm/ulaw/alaw codecs so far.
Tried disabling the echo cancelling as well.

Any suggestions will be greatly appreciated.

Regards,
Veselin

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Regards,
Veselin Kantsev
Campbell-Lange Workshop
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