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High Latency Networks


ToyotaHEAD
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Hello I am a NOC operator of a high speed internet over satellite network, and I am currently testing pbxnsip over the satellite network. Typically our network runs around 650-800ms of latency, but in high use times I have seen it spike as high as 2000ms. At any rate I am extreemly pleased with pbxnsip in a low latency network, but I am having a few concerns with the satellite setup.

 

First off, over the satellite does work very well! I understand the latency will be effectively transfered in the voice timming. However what I would like to address is the amount of time it takes a call to initilize (go though). Once the call has established and has been on the go for a few seconds everything just levels out and works perfectly. But the odd time it takes 5-15sec from when the call is placed, to when the other phone rings, to when the person on the far side hears the initial "hello". This being said most people would have hung up the phone thinking the call has failed to connect. Is there some settings somewhere that I can optimize either right at the server or on the VoIP phones themselves to help this problem? If it requires some settings to the actual satellite network I can even investigate there. Any suggestions or thought I will gladly entertain.

 

- Our test server is a windows pc 2GHZ, 2GB Ram. And is located on an Internet IP - (with no router)

- Our phones are typically the Linksys SPA921/SPA922 or the ATA SPA2102

 

Currently we are still demo'ing pbxnsip and our key expires in approx 2 weeks. I would like to see about ironing out this long startup delay before then, so I can get the boss's approval to purchase pbxnsip asap.

 

Thanks for your help in advance.

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I would say the problem you are experiencing is a combination of the SIP dialogue between the endpoints coupled with the delay over the satellite network. Obviously during an INVITE, the INVITE is then relayed to the remote party, they then negotiate, they then answer, the PBX then tells the remote party they've answered etc etc, you get the picture. If the average delay over the network is 3000ms, then each message is going to take 3s, you can see why a delay is possible during call setup.

 

In terms of fixes - I'm not aware of anything immediately obvious that would speed things up to be honest. The only thing I can think of is to play some kind of media until the remote RTP stream is received to give the impression they still haven't answered.

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If the average delay over the network is 3000ms,

 

I would say average latency is 850ish. But that still only translates to about 1.6 seconds, which is understandable, and moreso livable.

 

I'm sorry I forgot to mention.... This phenomon doesnt always happen either. It happens I'd say 1 out of 10 calls. And currently since this is only a demo, there is only the 1 active call on the server at a given time. I am just more curious if this is a possible hardware issue (Linksys phones themselves) or if there was some setting I have managed to overlook.

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I'm sorry I forgot to mention.... This phenomon doesnt always happen either. It happens I'd say 1 out of 10 calls. And currently since this is only a demo, there is only the 1 active call on the server at a given time. I am just more curious if this is a possible hardware issue (Linksys phones themselves) or if there was some setting I have managed to overlook.

 

Could it be intermittent packet-loss perhaps? The odd dialogue message lost and retransmitted?

 

You could maybe try the same setup on a LAN, that way you'll know if it is confined to the satellite network or not.

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You could maybe try the same setup on a LAN, that way you'll know if it is confined to the satellite network or not.

 

I started my tests with a LAN only network. Then mixed LAN / WAN. Both of which had supurb results. It is definatly a satellite network issue. I believe this issue is directly related to the higher latency. As a side note, we also run Fax machines but use some special hardware to facilitate it. This is again due to the latency.

 

Which sadly brings me back to the settings question. Is there no way to adjust pbxnsip or the phones themselves to make them more tollerant of a higher latency network?

 

Thanks again for all your thoughts.

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the 1 in 10 calls having delays may be the provider causing more delay , alot of the 2nd and 3rd tier providers tap into Level3 and Global Crossing, your timing may be affected by their least cost routing or moving you to another switch due to high traffic on their end , I would try a 2nd provider as well and compare results ,

 

if this delay is strictly handset to hendset registered to the pbx, then that rules out my theory since there is no provider variable at that point ..

 

yori

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the 1 in 10 calls having delays may be the provider causing more delay , alot of the 2nd and 3rd tier providers tap into Level3 and Global Crossing, your timing may be affected by their least cost routing or moving you to another switch due to high traffic on their end , I would try a 2nd provider as well and compare results ,

 

if this delay is strictly handset to hendset registered to the pbx, then that rules out my theory since there is no provider variable at that point ..

 

yori

 

This is a very good point. But at present I am just testing peer to peer though pbxnsip without going onto any pstn's. But like you say this also will add time once I get the funding to get a gateway in place. That is why I am trying to lower the timing between voip client as much as possible now.

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Ok I am somewhat confused now. I just learned that our current office Voip provider (a third party) is using pbxnsip. For giggles I plugged my office phone into one of our satellite modems. Even during an elevated latency over the satellite (1600ms) the call rang though and I was able to hear the person calling me withen 3 seconds. His server is located elsewhere on the internet.

 

Contrasting to that when i plug one of my test phones that utilizes my pbxnsip setup I am seeing calls connect anywhere from 3-15seconds. And by connect I mean person A is able to have a conversation with person B. Again once they connect all seems well. This makes me believe there is some trick I havent yet found. Like is there a way to spoof the invite request?

 

I'd be asking our current provder which ace up his sleeve he has played. However it is his service we are looking to retire; to bring inhouse. Thus he isnt willing to say what the trick is. He does say all the time though that he is "optimizing the network". What exactly is there to optimize if there are no settings to tweek?

 

I have double and tripple checked all QOS settings, as well as port filters. All is good.

 

Is there any other ideas?

 

Thanks

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