[WXC/VFX] WorldxChange VFX VoIP service

Forums › One New Zealand (including Vodafone, WxC, Farmside) › [WXC/VFX] WorldxChange VFX VoIP service

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BDFL - Memuneh
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#50815 2-Nov-2006 14:14

01EG: Registration form ask me for "Were you refered by an Existing WxC Customer?"

By Geekzone and this topic. ;)

But they want a person, don't know if existing customer can get any advantage from it but I can do it. :)

I need: title, first/last name and phone.

Hmmmm. They should accept Geekzone - it would be quite a few there now I believe...

maverick

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#50823 2-Nov-2006 16:22

Mattnzl:
Maverick - Is there any chance of capped calling rates being introduced on VFX?

We currently use Xnet for tolls, and get $1.85 capped national tolls for up to 2 hours.
Using VFX the same call could cost up to $6, so not as good a rate as the analogue line...

Thanks, Matt.

Hi Matt, sorry don't have an answer on this as it hasn't been decided yet

Yes I am a employee of WxC (My Profile) ... but I do have my own opinions as well

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nic.wise

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#50824 2-Nov-2006 16:49

Grant17:
01EG: Just got email:
The PAP2T, ID #90536, at $135.78, is in stock with our supplier in Auckland. If you were to order before 3pm, it would be delivered to you in Wellington the following day. All our prices include shipping and GST. Regards, Shane, GadgetPeople

:)

Take it quickly while you can. There isn't usually much stock of this item around the country.

$135.78 is not too bad. I think if you import from the US it will be more expensive.

Cheers,
Grant.

I think I paid $130 from Ascent. They have pulled it tho :( Looks like I got in at the right time!! :)

Nic Wise - fastchicken.co.nz

grant_k

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#50825 2-Nov-2006 16:52

nic.wise: I think I paid $130 from Ascent. They have pulled it tho :( Looks like I got in at the right time!! :)

Mine cost $87 + GST back in April...

01EG

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#50828 2-Nov-2006 17:37

Grant17: Mine cost $87 + GST back in April...

Just bought last one from Acquire for $95 + GST. :)

Fraktul

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#50832 2-Nov-2006 19:21

maverick:
nic.wise:
01EG:

Hardware, so I buy the PAP2T somewhere and make a settings by myself with your settings, is it right?

Does anybody really know how much traffic spend for 1 hour?

Thanks

OK, AFAIK the codec is running at 16K, which is 16 kiloBITS per second (I may be wrong here) so

16 * 60 * 60 = 57600 kilobits per HOUR

/ 8 = 7200 kilo BYTES per hours

About 7 MEG for an hour. Sounds about right, based on my knowledge of codec's vrs bitrates.

We have generally based it around

24Mb per hour for G729
76mb per hour for G711

So hence the reason we generally use G729 as the preferred Voice Codec

~7kbps for G729 including overhead (24Mb=25165824bits, (25165824/60)/60=~7000bps)? Thats amazingly good if true! If you remove Ethernet, IP and UDP overhead this would be more what I would have expected...

Mind you the overhead is dependent onwhere this is measured, for the purposes of WFX PPP accounting it sums to IP and UDP for overhead; but for your ADSL transport it sums to UDP, IP, PPP, ATM and ADSL overhead...quite substantial with small Layer 7 PDUs.

01EG

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#50833 2-Nov-2006 19:26

Fraktul: ~7kbps for G729 including overhead

what means overhead?

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grant_k

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#50834 2-Nov-2006 19:39

Fraktul: ~7kbps for G729 including overhead (24Mb=25165824bits, (25165824/60)/60=~7000bps)? Thats amazingly good if true!

Yes, that would be pretty much correct. G729's voice compression is very efficient but it requires significant computational resources to achieve it (usually a DSP chip).

G711 on the other hand uses a much higher sampling rate of 64kbps but doesn't do any compression so it needs minimal computing power.

I use a couple of VoIP boxes that have the older H323 protocol (forerunner to SIP). The Codec I use in those boxes is G723 with a sample rate of 6.3kbps. The audio quality is very similar to G729, yet the bandwidth needed is only 22kpbs as against 30kbps for G729.

Fraktul

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#50844 2-Nov-2006 21:18

Indeed, the point i was trying to make in a round about way was sampling rate does not equal or accurately give the actual data usage without taking into account overhead. Overhead varies a lot, particularly when talking about UBS and where you are measuring it.

Eg if you look at it between say the LNS and LAC its about as efficient as...well something very inefficient put it that way, I would estimate quite a bit less than 50% off the top of my head(lazy).

sbiddle

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#50873 3-Nov-2006 09:06

01EG:
Grant17: Mine cost $87 + GST back in April...

Just bought last one from Acquire for $95 + GST. :)

The PAP2's are in short supply everywhere, there has been talk of them being discontinued but I read somewhere that Linksys had denied this and are simply focussing on supplying them to network operators.

The SPA2100 is becoming very common in oz and has the advantage of being a router as well so you can get QoS on your internet connection if your existing router doesn't support it... Not that QoS is probably going to make a difference on a crap ADSL connection anyway! :-)

grant_k

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#50876 3-Nov-2006 09:33

sbiddle: ...so you can get QoS on your internet connection if your existing router doesn't support it... Not that QoS is probably going to make a difference on a crap ADSL connection anyway! :-)

After getting VFX installed yesterday, I sent a TXT to all my mates advising of the new phone number. Then received a lengthy call from my brother-in-law. During that 50 minute call, somebody sent me a large e-mail attachment and I started to get break-up on his voice. He could hear me OK, but not vice-versa.

This is strange I thought, clearly the Bandwidth Management functions in my router are no longer doing their job. After killing my e-mail program so we could carry on with the conversation, I made a mental note to investigate later.

What I found was that the "SIP / VoIP" QoS functions in my router were no longer identifying the VoIP traffic as such. Instead, it was being lumped into the bandwidth pool together with my PC.

I would say this has happened because VFX uses completely different ports than the standard SIP ones which iTalk uses. If you want to know what they are, you will have to ask Phil Moore because he doesn't want them posted on the forum for security reasons. Suffice to say, they are well outside the normal SIP range to avoid some routers out there that mess with standard SIP ports.

What this means, is that the QoS functions in your average router (if it even has them) will not allow you to prioritise the VoIP traffic, which is essential if you are to avoid call breakup on narrow bandwidth links (such as the typical 128kbps upload path of most ADSL plans).

The way I have got around this is by using a ZyXEL P334W wireless router which supports true Bandwidth Management functions. They are not that easy to get your head around, but I've managed to set it up now so that at least the incoming VoIP traffic is prioritised. I have done this by tagging all traffic to the IP address of my PAP2 as High Priority, whereas all other IP addresses receive Low Priority. For some reason, I still can't convince the router to identify Outgoing traffic from the PAP2, but that was never such a problem anyway. You just don't send large e-mail attachments while someone is using the phone. It's the incoming stuff you often have no control over.

QoS in general is a tricky area, especially so when non-standard ports are used!

maverick

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#50878 3-Nov-2006 09:56

sbiddle:
01EG:
Grant17: Mine cost $87 + GST back in April...

Just bought last one from Acquire for $95 + GST. :)

The PAP2's are in short supply everywhere, there has been talk of them being discontinued but I read somewhere that Linksys had denied this and are simply focussing on supplying them to network operators.

The SPA2100 is becoming very common in oz and has the advantage of being a router as well so you can get QoS on your internet connection if your existing router doesn't support it... Not that QoS is probably going to make a difference on a crap ADSL connection anyway! :-)

We will be supporting the the SPA2100 but primarialy the SPA2102 which is the updated new version it has 100Mb Ports as appossed to 10 Mb ports and newer hardware

Yes I am a employee of WxC (My Profile) ... but I do have my own opinions as well

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#50908 3-Nov-2006 12:59

Grant17:
QoS in general is a tricky area, especially so when non-standard ports are used!

If you don't understand QoS the easiest way is to simply set the IP or MAC address of your PAP2 (or phone) as a higher priotiry device. This will work in most cases and is a lot easier to configure.

It's not so much "non standard" ports that are used, it's the fact there are two parts to a SIP call - the standardised port 5060 or 5061 for the SIP call setup and then the RTP voice traffic which is seperate. There is no standarised post for RTP traffic so while some phones will use 5004 Asterisk for example uses 10000-20000.

If you're using a router that support L7 tags then both SIP and RTSP can be tagged which makes things slightly easier as you don't need to know the ports involved or tie up a MAC or IP address and the benefit that L7 filtering is a lot faster.

grant_k

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#50913 3-Nov-2006 13:38

sbiddle: If you don't understand QoS the easiest way is to simply set the IP or MAC address of your PAP2 (or phone) as a higher priotiry device. This will work in most cases and is a lot easier to configure.

That's exactly what I've done, using the IP address approach, and it does work OK for Incoming traffic where the Destination address can be identified by the router. To get it working for Outgoing traffic, I'm guessing that I would have to know the IP address of VFX's server, which will no doubt change from time to time and given this hassle, it's probably not worth the bother. Incoming traffic is the usual problem area.

sbiddle: It's not so much "non standard" ports that are used, it's the fact there are two parts to a SIP call - the standardised port 5060 or 5061 for the SIP call setup and then the RTP voice traffic which is seperate. There is no standarised post for RTP traffic so while some phones will use 5004 Asterisk for example uses 10000-20000.

Yes, I realise that. I guess that what I'm running up against is the relatively simple nature of the QoS functions on my router. Short of spending $1000 - $2000 buying a Cisco box, and then spending hours and hours learning how to configure IOS, I'm not aware of any other alternative.

Basically, the only QoS functions I have are as follows:

Select Service: VoIP (SIP), FTP or User Defined

the VoIP SIP option used to work with iTalk in that it would pick up the VoIP traffic. Since switching to VFX, it no longer works.

Bandwidth Budget: 10kbps to Full Speed of the Destination Port (LAN, WAN or WLAN)

I find that reserving 30 or 40kbps exclusively for VFX is sufficient to prevent breakup on incoming calls, regardless of other activity on the downstream link.

Priority: Low / Medium / High

Obviously, I assign High priority to VoIP traffic and low to everything else.

Destination Address / Netmask / Port
and
Source Address / Netmask / Port

By using the Netmask correctly, I have managed to specify either one IP address only, or a range of addresses.
However, the same flexibility is not applicable to Ports (you can enter one only). In order to cover the range of ports used by VFX, I would have to make potentially hundreds of entries, which is beyond the capabilities of my router.

Protocol: TCP/UDP/User Defined (with a Protocol Number -- whatever that is?)

Again, this parameter is not much use unless you make a separate entry for each and every port in use by VFX.

sbiddle: If you're using a router that support L7 tags then both SIP and RTSP can be tagged which makes things slightly easier as you don't need to know the ports involved or tie up a MAC or IP address and the benefit that L7 filtering is a lot faster.

It doesn't look like my router supports L7 tags. My older VoIP system using H323 protocol allows me to set the "IP Precedence" to a number from 0 to 7. I set it to 7 (being the highest priority) but to be honest it didn't make much difference. The only way I have managed to successfully stop call break-up in the presence of heavy LAN traffic, is to use the Bandwidth Budget settings on my router.

If you have any suggestions as to a better router I could be using, please let me know. I'll find out the cost and have a look at the Technical Manual to see if it will do the job.

Cheers,
Grant.

sbiddle

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#50931 3-Nov-2006 14:58

I used to use a WRT54g with Sveasoft firmware that supports L7. QoS works extremely well on this - simply set a rule to filter VoIP traffic (catches all of the main protocols), and set your upstream & downstream speed and it works pretty well.

My current setup is a HP NetConnect server running ClarkConnect 4.0 that also runs my Asterisk PBX on it and CNet managed switch that supports QoS. I have ClarkConnect tagging all incoming VoIP packets and have priority given to the associated UDP ports (10001-10050 in my system) as well as QoS set for my upstream & downstream speed. This gives me perfect quality both incoming and outgoing over my TCL cable connection.

The whole QoS issue can be very complex, you only have to read some of the online forums to realise this. The downside is that unless you have QoS enabled you will suffer degredation of voice service if you try and use the interweb at the same time as talking on the phone without it.

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