VOIP OVER IPSEC GLOBAL VPN

[NETING]

MY question is as follows. We have site to site IPSEC tunnels over the Public internet spanning the globe to each of our sites. Domesticaly in the United States we have one provider. Internationaly we use another provider. We will be rolling out VOIP shortly. My question is will the performance or quality suffer because of the fact that we will be running VOIP over the public internet as well as having two different providers. QOS can be done but after a site is left it will be best effort over the internet until it reaches the other site. 3 offices in the UNITED STATES...2 in the UK.... 2 in Hong Kong...

Thanks in advance.

 

[AlexIT]

You may want to set up monitoring for the IPSEC connections. See what the traffic levels are and if there are any bottlenecks or time dependencies. There should be no real difference between IPSEC VOIP across one provider vs across two. We have not implemented VOIP yet, but have a test case running now between two locations in the US (different providers.) Our only issue is the odd echo/lag that happens ocassionally. Otherwise its very close to a standard call.

Alex

 

[gacollier]

Neting,

Not sure which VOIP solution you're thinking about, but I'm currently using an H.323 solution over a 4 site LAN to LAN VPN. Quality is excellent over all points with the exception of one office that's currently using a wireless Internet provider (rural area). All sites use different providers and different technolgies (T1, ADSL, Cable, and Wireless). All connections have less than 100 ping times to each other with the exception of the Wireless point (typically all over the place, 100 - 3000 ms). I would say if you can get s fairly consistent ping you should be in business.

Good luck.

 

[NETING]

Guys thanks for the feedback. I dont think we would have a problem here in the UnitedStates. For the same reason all pings are under 100ms. But when going over seas to the UK the ping time is 120ms. Provided we stick with the one provider. the problem is this response time to the UK would increase becuase it would have to go through network access points inorder to do a carrier exchange and then go to our UK facility. Domestically I am not worried about VOIP because we are using it and it works.

Hope i make sense.

 

[AlexIT]

Neting,
"But when going over seas to the UK the ping time is 120ms."

I suspect you mean 1200ms?

Gacollier,
Does the quality issue result in bad lag/echo situations as we've seen? When looking I've seen 5000ms ping time...I wonder what the max before becoming audible would be (i.e. is 500ms still good?)

Alex

 

[NETING]

AlexIT,

I am in the Northeast and the ping times to the UK are about 120-130ms average. I guess I was wondering if this was acceptble. We have a site to site tunnel to the UK and ping looks as follows 111ms...123ms....126ms...140ms...

To our Hong Kong office the average ping is about 330 ms..

Good question on the one above (i.e. is 500ms still good?) Also doing VOIP over VPN is it true that staying on one carrier network is better.

As always the help is appreciated.

Neting

 

[AlexIT]

I do not know about the single carrier idea, we have mulitple carriers on the test link (and will have the same in production.) I can't see where there is much difference with a normal call from Verizon piece of ATT's network to USWest piece of ATT's network VS VOIP VPN call with same pieces of ATT network...its all routed packets after all.

Our test connection sounds fine at 100ms, but when it gets bad I check and see pings over 1000ms. I have no idea at what point you begin to hear problems. I might suggest a test with a re-routed connection where you can talk to yourself (like a call from the US to UK site with call forwarded back to US site...) Then you could see if two legs at 120ms is going to cause audible problems. With our test case being carrier1 to carrier2 (wireless) all in the US my results will vary from yours.

Alex

 

[gacollier]

Alex,

For me, the sound is choppy, cutting in and out. Again, I'm using H.323 with the Microsoft CCITT G.711 A-Law CODEC @ 64000 bit/s bitrate. If you've got a client that supports it you might try a 48K bitrate, or a G.723 codec (supports bitrates 5.3 - 6.3 Kps). Sound quality will drop, but it's worth a shot. Not sure how SIP would perform.

Good luck.