I'm looking for advise on troubleshooting QOS problems. After much
searching and reading online (Google, Voip-Info Wiki, etc.) I don't feel
any closer to finding the right tools to solve my problem. Any info you
would like to share would be much appreciated, and I'm sure the thread
will server others in the future.
The problem :
-------------
I'm having intermittent problems with the audio cutting out on calls. At
the same time the audio problems occur, I often see these in the "full"
log :
Received iseqno 122 not within window 123->123
These range from sounding like bad cell phone calls, to the audio track
cutting out in one or both directions for up to 20-30 seconds.
I also see dropped calls that seem to be a result of the IAX connection
going away.
The environment :
-----------------
I've got an * server located at a data center with good connectivity, 10
hops to my IAX provider, and ~34ms ping times. They (IAX provider) use
Cogent which concerns me a bit, but I'm not ready to jump to conclusions
just yet.
My IP phone is connected via "enhanced" DSL (static addresses, no PPPoE)
and I'm 12 hops away from my * server. My DSL provider has direct
connectivity and peering agreements with the data center my server is
located in. I've set QOS priority on the LAN port (Linksys router) the
phone is connected to, and I've dropped the MTU to 576 as suggested for
lower speed links. (1.5Mbs/384kbps in my case). Both these changes
seemed to make an improvement over previous calls. Currently I don't
believe the bulk of my problems to be between the phone and the *
server. testyourvoip.com tests consistently show a 4.4 score (the
maximum for ulaw) and rarely shows errors.
Ulaw is the codec used for both the SIP calls and IAX trunk.
What I'm looking for :
----------------------
I'm trying to determine the cause and location of the problem between my
* server and the IAX provider (and possibly my IP phone), and see what
if anything I can do to reduce the occurrence of these drop outs. I'm
looking for a couple of things :
1. A method of monitoring RTP/IAX traffic QOS at the PBX in real time.
2. Tools that might be used to determine the location of the problem.
I.E. An RTP/IAX "traceroute" tool.
What I'm hoping to find is something that either integrates directly
with *, or captures live RTP/IAX traffic and provides real time
statistics on calls.
What I've found :
-----------------
I saw Telchemy's VQMON_EP product, but it's unclear how it would work
with Asterisk. Many other companies in this market seem to leverage off
of Telchemy's products.
http://www.telchemy.com/partners.html
http://www.voiptroubleshooter.com/tools/voiptr_tools.htm
All of the products above seem to be aimed at large enterprises with
deep IT pockets. I wouldn't mind ponying up a reasonable sum for a tool
that does the job, but I lack the time to thoroughly evaluate everything
that may be out there.
I haven't found much on the open source front. I've seen "Windows RTP
Quality Monitor" which might be useful, but it's beta and hasn't been
updated in over a year.
It seems to me that Ethereal might be integrated with a graphical tool,
and if nothing else provide postmortem statistics on a phone call.
Request for comments :
----------------------
What are people using to troubleshoot these problems? What commercial
software works for you? What open source projects are you using? Do you
have suggestions on projects that might be glued together to provide
this functionality?
Thanks in advance.
Chris
_______________________________________________
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
