Definity VOIP Quality Question

[STF26]

I have a client that has 8 remote sites with 2-3 IP Phones in each site. There is a Cisco Router in each site with a dedicated T-1 back to the main office where the Definity G3Si is. The Core Router is also a Cisco router with a single t-1 connected to it. There are 3COM superstack switches in the remote sites. There are only about 6 devices on each remote network 2-3 computers and two 2-3 phones. The computers are plugged into the phones. There are many people and vendors involved in the project. The routers are managed by ATT so you cannot get into them the switches are managed by the company so we can get into them. I said in my opinion: Do not make things more complicated then they have to be. I do not think that VLANS and fancy QOS are required. The amount of traffic being passed to the remote sites is so small it barely registers. (Terminal Services Traffic for three computers).

The problem is that they are having issues with dropped calls and quality. When a phone is placed at the core site where the G3si is located the phone works great. So I think that would rule out a problem with the pbx (medpro/ clan).

Any advise would be much appreciated.

 

[NX01]

Can you do a list trace st XXXX and post the results. XXXX being the ext of one of the problem ip phones. Are these hardphones or softphones? Also, are all the phones on the same network region? Are you using g.711 codec?

In the future everything will work...

 

[STF26]

All of the remote sites have problems with quality and dropped calls. They phones are hard phones. Now that I think about it, if all of the remote sites are having problems I would have to point my finger back at the bottle neck at the core router. Hmmm.... Yes they are using the g.711 codec. I cannot run commands on the PBX but someone at the site may be able to.

 

[NX01]

Are there any gateways at the remote sites? Have you tried setting the phones to 10mbps/full? Set your medpro and clan to the same.

In the future everything will work...

 

[STF26]

No VoIP Gatways in the remote site. A very simple network. Maybe there is a problem with port negotiation on the router / swiches in the remote sites. We have checked the statisitics in the remote sites on the 3 com switches and they look pretty good. I am very curious about how much traffic is passing on the core router. ATT needs to allow us to gather SNMP data from the router for analysis.

 

[Rod25]

Hi,
How often do they drop? I would run a continous ping to the CLAN that the IP Phone is registered to and another ping to another device at the central location and make a call. If call drops and you get icmp relies then you need to look at the WAN link configs.

Rod

 

[NX01]

I think the default for icmp is every two minutes. I changed mine to every 10 secs. Also look at how the nating is taking place. You may have to setup a seperate network region for EACH phone with a limited UDP port range. i.e. Two to three ports for each. Then you can port forward those ranges to the specific ip hardphone.

In the future everything will work...

 

[wildcard100001]

STF26,

Even with minimal network traffic, QOS is still required in a VOIP environment to be successful. You will have problems as you describe if you don't use it. Network "overhead" will cause issues without QOS. Apply QOS to the phone ip address range in question and the routes points they take, not necessarily the whole Vlan. Use QOS even you have tons and tons of bandwidth. The vast majority of VOIP issues come from not having QOS implemented.

I base my knowelege on experience and VOIP install horror stories.

Thanks,

Wildcard

 

[STF26]

Thanks for everyone's insight on this issue. After talking with Avaya, ATT and 3 com the client is going to implement some voice and data vlans in the remote sites. The QOS is supossdly setup on the wan routers so lets see what happens. I will post back once I get some additional information.

STF