[WXC/VFX] Latency on call forward

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aarond

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#22886 11-Jun-2008 11:28

Now that I have got call forwarding and follow me working on my own Trixbox setup at home, (see my post here http://www.geekzone.co.nz/forums.asp?ForumId=65&TopicId=22871) I have noticed that there is quite a bit of latency on a fordwarded call. This is only when I have two active calls, e.g. an inbound call then forwarded to an external number. The latency can be up to 5 or 6 seconds.

Do others find this? Is this because the server I am using is not grunty enough? I'm using ulaw and alaw, not G.729 - would the latency be less if I used G.729? Any other advice to minimize the audio latency?

Aaron

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#137273 11-Jun-2008 11:45

What are the specs of the server? And your internet connection?

aarond

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#137280 11-Jun-2008 11:52

Its a 2.5GHZ Celeron processor with 1GB RAM, running WHS and Trixbox in a VM. Most of the time its just serving up media to XBMC, so nothing computationally excessive. I realise that the VM would cause it to run slower, so certainly I am thinking about running it on a dedicated server if thats going to give me significantly better performance.

Also, internet is XNet River with VFX - so only 128k up. I am planning on moving to Fusion once I am satisfied that it is all running well without any issues. I thought a full upstream connection might give me the best performance improvement in terms of handling multiple simultaneous connections.

If forwarding is working well for you, I would be interested in hearing about anyone elses setup specs.

Aaron

sbiddle

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#137293 11-Jun-2008 12:27

VM + trixbox = bad.

It's also not surprising you get getting audio issues if you're only on 128kbps - ulaw/alaw both use around 87kbps in each direction for a call.

aarond

33 posts

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#137300 11-Jun-2008 12:47

Thanks for the info. I didn't realise it was as high as 87 kbps. I will try something else, what sort of bandwidth does g729 use?

I am willing to put Trixbox on its own dedicated server, I just want to try and do everything I can to get it running in a VM first. More efficient use of the hardware I've got. From what I have read, Trixbox + VM = bad only for timing critical applications such as Meet Me, which I have no intention of using. Other websites I have read show people running it quite successfully on WHS without issue, although generally they say that just after they have installed it, so I have no idea what kind of problems they run into further down the track.

Aaron

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#137327 11-Jun-2008 13:58

Why don't you just switch to the Flood plan? Sure a few more $$ in the short term, but at least it will allow you to test properly before moving to Fusion. Also, if your server is under powered, going to G729 will make it worse. G729 requires more power to code / decode the audio, due to the compression.

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aarond

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#137329 11-Jun-2008 14:02

A couple of newbie questions here.

I (naively) tried to add G.729 codec by changing my trunk to allow=g729&ulaw&alaw. I know that g729 is licenced, but I was curious to see what would happen. I thought that at the very least, it would not be licenced on my machine and it would fall back to ulaw. However, my incoming calls instead just hung up. Take it out, and it is back to working again. Why is this?

Second, what codecs does VFX support? I read some articles saying that G.726 was a good choice offering low bandwidth requirements and reasonable voice quality. Am I able to use this?

Thanks
Aaron

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aarond

33 posts

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#137330 11-Jun-2008 14:04

Great idea on the flood plan. I hadn't thought of that. lets me test things out with the same network connection I would have on Fusion.

Thanks

sbiddle

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#137334 11-Jun-2008 14:09

aarond: A couple of newbie questions here.

I (naively) tried to add G.729 codec by changing my trunk to allow=g729&ulaw&alaw. I know that g729 is licenced, but I was curious to see what would happen. I thought that at the very least, it would not be licenced on my machine and it would fall back to ulaw. However, my incoming calls instead just hung up. Take it out, and it is back to working again. Why is this?

Second, what codecs does VFX support? I read some articles saying that G.726 was a good choice offering low bandwidth requirements and reasonable voice quality. Am I able to use this?

Thanks
Aaron

Asterisk does not have a g.729 codec installed so if you try and use it it will just fail as it can't do any transcoding, just passthru.

You can download a free G.729 codec from here, installation is very simple - just download the .so file and reboot.

http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/

aarond

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#137338 11-Jun-2008 14:17

Thanks for the info sbiddle. I will follow up on that link, but for now I think I will move to Flood first.

Has anyone noticed that info on the various broadband plans are no longer accessible through the XNet website? Am I blind? I click on Internet > Xnet HSI Broadband in the menu, and it tells me that I can do pay per meg or Torrent, but I can't find any links to the various Stream, River, Flood plans? I just want to know the price, and check for other conditions. Well, I guess I will have to ring them.

Thanks
Aaron

cisconz

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#137339 11-Jun-2008 14:18

Try seting the call to not send RTP via your asterisk Then only SIP signaling will be send down your line. All voice traffic will go WxC - WxC.

Hmmmm

coffeebaron

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#137343 11-Jun-2008 14:22

Flood $49.95 per month + data
[edit]
http://www.xnet.co.nz/hsi/pricing.shtml

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aarond

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#137344 11-Jun-2008 14:22

Rung them up, changed to Flood (Rebecca was very helpful, as usual), will take up to 2 days to take effect.

Now just have to explain to my wife why I have done that without consulting her! And just when I had convinced her VoIP would be cheaper...

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#137346 11-Jun-2008 14:26

aarond: Now just have to explain to my wife why I have done that without consulting her! And just when I had convinced her VoIP would be cheaper...

You are just "beta" testing, so that when you go to Fusion, your wife will have no problems with the service.

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aarond

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#137350 11-Jun-2008 14:34

Oh, of course... she shouldn't be questioning my motives!

Everything I do is ultimately for her benefit

cisconz: Try seting the call to not send RTP via your asterisk Then only SIP signaling will be send down your line. All voice traffic will go WxC - WxC.

Like any good geek, I will go and do some reading to understand what you just said. But maybe in the meantime you could help me out by explaining this in more detail. What's RTP? How would this work? Does this mean audio for inbound calls would still come via asterisk, but when the call is being forwarded you can somehow instruct WxC to do the forwarding itself? On a call by call basis? That is, changes depending on whether I accept the call on my mobile vs at home?

cisconz

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#137373 11-Jun-2008 16:02

aarond: Like any good geek, I will go and do some reading to understand what you just said. But maybe in the meantime you could help me out by explaining this in more detail. What's RTP? How would this work? Does this mean audio for inbound calls would still come via asterisk, but when the call is being forwarded you can somehow instruct WxC to do the forwarding itself? On a call by call basis? That is, changes depending on whether I accept the call on my mobile vs at home?

RTP is the audio part of the call. i.e. Sip runs on Port 5060 but RTP can run on any port as that is established on a per call basis.
A call at G711 takes about 86k for the RTP and 1k for the SIP Signaling.

Normal Sip Call

This ^^^^^^^^ is how a normal call comes in to your Asterisk Box.
This senario uses about 87k per call.

Redirected Call

When you redirect a call if you deselect RTP Proxy then the call will go like this ^^^^^^^^^^^.
This senario only uses about 2-10k per call as all the audio gets dropped back to WxC.

So this is on a per call basis and for example if it calls your home phone then out to you cell the RTP will come through for the first part of the call (local extention), then drop back to WxC for the divert out.

Hmmmm

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