Expert Needed - Call deteriorates after 2 mins in

daschenbrener

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#1
Am an able to make calls fine, but after aprox 2 mins into the call the other end indicates the call is choppy and then eventually cannot hear and I have to end the call.
The call is going over an ITSP.

And it makes no difference if the call is incoming or I dial out.

I can hear them perfectly fine and without any break or anything.

I have got my internet provider to come and diagnose my feed and they indicate it is fine.
On elastix 2.0
aterisk 1.6.2.10

Not too sure where to start on this one. and direction will be appreciated

Thanks

David
 

andyshawn

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#2
These are my thoughts...

This sounds like an upload/bandwidth problem. Since its the other end that complains that the call starts to deteriorate, then i am guessing that your upload might be limited or your ISP is doing some sort of traffic shaping.

Does this happen with every call you make to different individuals? If you make a call to a device on the same LAN, does the same thing happen?
 

daschenbrener

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#3
No it does not happen on the lan side.

I am on cable with apparently 1mb upload.

But never thought to ask if they are limiting my upload.
 

andyshawn

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#4
Its possible that they are dropping your packets. As you know voice traffic is 'very' sensitive to delays and it needs the required bandwidth in order for the packets to arrive in order and ontime. Since you are using cable internet, this is a shared medium and your upload bandwidth of 1mb is not guaranteed.
Have you tried making calls late at night and early in the morning, where i'm assuming the network traffic is low?
 

daschenbrener

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#5
It did not seem to make a difference. This is also with 2 different ITSP also.
 

daschenbrener

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#6
getting off with my ISP, they indicate all is good and signal and up/down speeds are consistent.
Could it be a deteriorating switch I have in my network?
It is a netgear gsm 7324 managed layer 3 switch, wide open without any management on.
 

andyshawn

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#7
I ruled out your network devices as the problem, because you said calls it works good on your LAN. Where is the server located? Is the server overloaded? Do you know how much memory is in the server?
 

daschenbrener

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#8
3GB of Ram, 2.8ghz HT intel processor, Sata drives. only 7 users on the machine
 

andyshawn

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#9
Ok then, the possibility of the server being overworked is eliminated then. I can't think of anything else at the moment. Are all other users experiencing the same problem, if yes, then it might be a problem with your router.
 

daschenbrener

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#10
Yes they are.
I am using pfsense for my routing. on a p3 800 with 500 mb of ram
 

andyshawn

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#11
I think thats where the problem is. I assume you are using the firewall feature and your firewall maybe doing some packet filtering/checking which slows up the process. Is it possible for you to use a regular router just to eliminate pfSense as the problem?
 

daschenbrener

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#12
Question being, what kind that can handle the traffic?
 

andyshawn

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#13
I'm think any other router, like a linksys wrt54g2 or a dlink router. I just want to change your gateway for a minute and do some testing.
 

dicko

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#14
JM2CWAE

pfsense is in many ways more competent than a "regular router"

That daschenbrener is using a cable company, is in itself suspicious, I believe he is in Jamaica, but here in the US, there are many class actions against the "cable companies" for traffic shaping, many here offer a voip service and usually not SIP, it is not in their commercial interest to allow SIP, One of my personal experiences is that Time-Warner will disrupt your rtp traffic for 10 seconds every 5 minutes, they of course deny it.

you can see the flow of data by issuing

rtp debug ip <your provider's ip>

at the asterisk CLI

and analyze the synchronicity of the traffic flow as the calls degrade.

(for postpartum forensics, the same data will be timestamped and available in /var/log/asterisk/full)

dicko
 

daschenbrener

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#15
I am in regina, sk,canada. My isp is access communcations.
They claim they are not traffic shaping at all.


Also when you indicate

rtp debug ip <your provider's ip>

do you mean the ip address i am using for my voip server?
 

dicko

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#16
My apologies I copied and pasted the wrong IP address, Regina is much colder than Jamaica.

but no, the connection is between your external IP and your VSP so that would be the ip of your Voice Service Provider's server.

They ALL claim they are not traffic shaping, that does not mean the aren't. You will have to prove it to them, they will then of course deny it again.

Have you set up the QOS on the pfsense correctly?


good luck

dicko
 

daschenbrener

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#17
for whatever reason in my version of asterisk, 1.6.2.10, rtp debug ip does not work?
 

dicko

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#18
I believe it does, you are are probably just doing it wrong.

if you are confused and have only one phone call at a time

rtp debug

should suffice.
 

daschenbrener

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#19
the rtp set debug on works, the rtp set debug ip, does not, command always fails
but that is fine, i can track the log on my calls.

So what is it that I am looking for in the log?
 

dicko

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#20
when a "Sent RTP packet" is not immediately followed by a "Got RTP packet", watch the seq field.

if that doesn't happen impeccably you are hosed,

If you are using asterisk 1.6 then

rtp set debug on

but

rtp set debug ip <the ip address>

should work just fine. replace <the ip address> with the ip address you are interested in e.. 123.123.123.132


(my advice as of right now is, don't use Elastix 2.0.? yet)
 

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