Avaya IPO latency and echo

[Jimbo2015]

I am having real troubles with one of our customers where their SIP calls have latency and echo. The main issue being echo.

The SIP provider is sure it's nothing to do with them as the issue is affecting many destinations and their other customers are not complaining.

Their IT company says the phones are on their own network and there is nothing on their side which they think is causing this.

Is there anything on the Avaya that has been known to cause this in the past and is there anything you can do to limit it, either through settings on the handsets or on the system config?

Thanks in advance.

 

[jamie77]

Latency on the internet curcuit.

What curcuit is this? Can you get wiresharks to check for latency?

Jamie Green

[bold]A[/bold]vaya [bold]R[/bold]egistered [bold]S[/bold]pecialist [bold]E[/bold]ngineer

 

[amriddle01]

Network or internet connection, 100%, always is.
That's how SIP works, by traversing their network and then the internet, if you get audio issue that's where you look

 

[Jimbo2015]

Ok thanks it's a 100mb leased line using a Cisco 3750.

Could the internet also be causing echo?

Their IT don't have a clue about SIP so is there anything you think we should ask them to do on that router? I have asked for SIP ALG to be disabled but whether it is I don't know.

I will try using wire shark when I go there on Wednesday.

The issue seems to be worse since their call volume has increased.

Also would enabling or disabling direct media path help at all?

Thanks

 

[Jimbo2015]

P.s some calls just randomly drop too although this is rare and not nearly as often as the echo issue

 

[amriddle01]

Mirror the IPO port, then wireshark trace. Wireshark can play the audio it captures back, so capture a test call. If you hear jitter/echo on the outbound during the call but not in wireshark ...you know it left the system fine. Also if you hear jitter/echo on the inbound you know it's also happening before it hits the system, really just proving it's not the IPO (which we already know).
Then you can hand it over to their IT, you can't troubleshoot their network/connection, if they can't they need to learn, your work there is done

 

[amriddle01]

I use this to test jitter/packet loss, it's not perfect but tends to give a good indication

http://www.pingtest.net/

 

[RodneyMcSnow]

Perform a tracert to the sip provider and see what the ms response is.

Also, is there a sip firewall (Session Border) being used between up office and internet provider?

In the ipoffice, is NAT enabled under system and LAN port?

 

[Jimbo2015]

Thank you for the help here.

There is no SBC in place. NAT is not enabled for either LAN 1 or 2.

I will go to site and use wireshark in the next couple of days.

I just connected through teamviewer and used pingtest.net. It showed jitter was 1ms and the ping was 16ms. There is no one at the office now using the connection so maybe this would be worse during the day.

Direct Media Path is not allowed for most extensions as this was recommended previously with a similar issue.

It shows jitter on quite a few extensions on system status.

Also I thought echo was caused by analogue to IP convesrions but there is no analogue in this case.

What would you suggest I ask their IT provider to do? I was thinking I should ask them to increase the bandwidth they have given to the telephony network on the leased line. Also I do not believe their is any QOS or VLAN in place so was thinking of suggesting this?

Many thanks for the help here.

 

[gotchaback]

Jimbo what codec are you using on the sip line, G.711U is what will allow you the most margin for error. how many talk paths does your customer have and if you have a 100mb up that should more than enough.

 

[Jimbo2015]

G.711 A and U law
G.729 and G.723 are all selected.

There is 100mb but I am not sure how much of this has been assigned to the telephony portion of the network.

Thanks,

 

[Jimbo2015]

Please see the config attached. If you could all please tear this config apart and let me know everything I have done wrong it will be appreciated

 

[gotchaback]

If it were me i would find out how much bandwidth i had to use and how many talk paths i had to support it; i would also set my codec for that line to g.711u only.

 

[Jimbo2015]

Their IT has suggested the following but do you think this could pose a security issue?

'As the phone network is separate, would you suggest and consider making the WAN interface of the PBX to be Internet facing rather connecting through the ASA?
This would bypass the ASA altogether.'

 

[RodneyMcSnow]

For testing purposes you could but you can apply a firewall on Lan1 also.

In any event I [highlight ][/highlight]would go ahead and run the sip for an hour or two to rule out any internal firewall issues and see what the results are at that point.

Make sure you change your routing, you also need the public Internet IP the subnet out and gateway you're going to be pointing the sip trunks to across the internet.

Make sure to disable h423 and sip provisioning on the LAN tab once it's up and running.

By the way he way if you perform a constant ping from a command prompt to the public up (ping xxx.xxx.xxx.xxx -t) while your performing this what are the MS response times you are seeing from the public sip provider?