Canadian Telco Telus Moves All Call Traffic to the Net

justice0x1 writes "An article on the Tornonto Star about Canada's Telus making a large scale motion to move all call trafic over to IP caught my eye today. 'Telus will become the first dominant phone carrier in North America to make the risky transition, a move much talked about and which Telus will make happen on a dramatic scale.' Since I work in the Telus Internet Service department, it will be interesting to see exactly how this new technology fares. Seems almost premature to me, but I guess it's all or nothing with telecomunications these days; you need to get an edge on the competition somehow. Why not start by moving youre entire long distance network over to IP?"

Comments

Re:IP != The Net

[8282now]

Just because it's VoIP, it doesn't mean it has to flow over the "public" IP network. They can simply point the VoIP over their own network and still see massive savings.

For the VoIP/PSTN imparied, VoIP traffic can be optimized (that's compressed) down to 8K (or more, but a cost of clarity) as opposed to 56K for traditional circuit switched telephony traffic.
And yes I have worked with the technology.

Re:Bandwidth?

[PerlGuru]

I use Vonage VOIP for my home phone and they require 90kbps both ways in their default config. They have an option to reduce the call quality, and thus the size. 90kbs is about 2/3rds the way to standard phone quality comming from cell phone quality, if that makes since to you.

Re:Not the public Internet (I hope)

[n3k5]

I very much doubt that they'll route calls over the public internet.

You _doubt_ it, your parent says _probably_ not. Well, they're _definitely_ not connected to the Internet in any way, which becomes apparent if you just read the first few paragrphs of the article. The /. headline says something different, of course. Great job, timothy!

Re:Bandwidth?

[AmunRa]

In europe, traditional circuit-switched telephone networks use channelised E1s, which basically have 15 64kbit/s data channels, along with a 64kbit/s control channel, giving a 2Mbit/s channel.

Using H.323 (the main VoIP standard) you can choose from a number of compression codecs, from 32kbits upwards, although there exists a couple of codecs which sound pretty good (coparable to a average analogue phone line) at 64kbit/s, to you can make the transition without effecting you capacity.

I was involved in some trials here in the UK, and (IIRC) we were using about 90kbits, and that was _crystal clear_ - the thing you most notice is that when no-one is talking, it is actually *silent* - you don't realise it until you try VoIP howe much noise is in a normal analog call.

This is fact one of the advantages of VoIP, you can have in effect variable bitrate encoding, as if no-one is talking, then you don't have any data to encode!

Re:Bandwidth?

[jcdr]

Nope!

E1 is 30 B channels at 64kbits/s + 1 D channel at 64kbits/s for signalisation plus 1 synchronisation and misc operator stuff at 64kbits/s. So E1 is 32*64k=2048Mbits/s.

Some operators have services for 10, 12, 15, 20 or 24 channels depending of the offert but this is simply a limit of concurrent B channels open in a 30 B channles capacity E1 line.

Hold the phone.

[FreeLinux]

As a few posts have already descibed, VoIP is not the same as voice over the internet. The VoIP that Telus is using is actually being carried over dedicated data circuits which offer, not only high speed (OC-12 +), but also end to end Quality of Service control. Indeed, these dedicated data circuits may well carry some internet traffic as well, but Telus is able to run the voice traffic at a higher priority than the less time sensitive data.

In fact, this is already being done by several carriers including Sprint, MCI, Intermedia, Verizon and probably others.

I have also installed numerous private networks utilizing Nortel or Cisco equipment to carry VoIP over dedicated private networks, usually frame-relay WANs. These VoIP calls are 100% reliable and are perfectly clear.

In two cases standard internet connections (cable, xDSL, frame-relay) were used to carry calls between several different offices in the United States, Canada, Europe and Mexico. These connections are not always as clear as those running on private WANs but, they have proven to be 98% reliable and are indistinguishable from regular land lines, in terms of clarity, 85% of the time.

Re:I guarantee this will fail

[EQ]

Try learning more modern stuff than whatever antiquated VoIP crap your company was using.

As an example, I was working with VoIP stuff 3 years ago with Cisco, T-Mobile and Level 3: we had a project that took a GSM signal, put it onto a local (in building only) network, and from there to VoIP and onto the Cisco Call Manager system. This allowed the cell phone to be your internal company phone extension while on campus, but automatically switch back when off campus. We faced many of the same problems you cited but all were solved by the end of the "beta" test.

Also we deployed a nation-wide softswitch infrastraucture that allowed us to hook Cisco VoIP phones at any point in the network and make toll-quality calls out from the main PBX back in Broomfield. This saved the company from having to drag a copper pair to each and every little regeneration site onthe fiber right of way, which is not an inconsiderable amount of cash when you figure the extent of Level 3's network.

Update your knowledge base. IP is starting to be picked up now that the private IP networks Like Level 3, Genuity, plus parts of AT&T and Sprint have worked out the problems. Draft Martini (Read the IETF documents) has been delivering standard phone services like ATM and FR over IP for a year. And Level 3, where I used to work, has had an all IP infrastructure for several years now, using the old XCom (Now VIPER) sofswitch. Want to know where those cheap long distance calling cards are coming from? They probably travel over Level 3's all IP network, converting at the edges on the softswitched architecture. And a good percentage of Worldcom/MCI long distance moves on Level 3's all IP network as well.

Secondly, the VoIP mentioned here is primarily in the backhaul, behind the CO switches (RTFA again!). For Inter and Intra-LATA carriage, VoIP is highly reliable, and much less expensive than the TDM stuff.

As for the "silence", RTFA. Seriously, they solved this one a long time ago with a little echo back on the reciever side, as well as "comfort noise" from the IP switch. This technique goes back several years.

Seriously, you need to learn more - and RTFA, because you are "mistaken about a great many thing"....