eircom to launch voip service..presently

Sponge Bob

you must be a customer on their bb, have no prices so ring them

1890 260 260

fatherdougalmag

Courtesy of sirlinux on the VOIP forum <linky>. Googling will get more info but more linkage. They use a Zyxel Prestige P2001 ATA that you plug into your existing Eircom router.

I wonder how BT (well, UK anyway) will react as it seems they already use the marketing term "BT Broadband Talk".

towel401

If you can't beat em, join em!

I expect this service to be more expensive that regular old VoiP only providers.

will they be using proprietary equipment/protocols, or SIP?

watty

Why don't they simply offer cheapr line rental and phone calls. Same network and more reliable. VOIP is for operators that don't have copper network.

matrim

towel401 wrote:

If you can't beat em, join em!

I expect this service to be more expensive that regular old VoiP only providers.

will they be using proprietary equipment/protocols, or SIP?

It will be SIP based.

Peanut

watty wrote:

Why don't they simply offer cheapr line rental and phone calls. Same network and more reliable. VOIP is for operators that don't have copper network.

To try to sell more broadband I imagine... as far as I remember there is no growth in the fixed line business so they have to focus on broadband & related services to increase revenue. (apparently.)

gerryo

watty wrote:

Why don't they simply offer cheapr line rental and phone calls. Same network and more reliable. VOIP is for operators that don't have copper network.

Indeed, it does seem a strange move, they already have the necessary infrastructure for voice calls, so why the duplication?

I'm wondering if they assume line rental is the only guaranteed income they have for the future, keeping customers on Eircom lines by any means necessary is the new business model.

Offering Voip only makes sense if you tie customers into Eircom Voip, otherwise they'll nip off to blueface, freespeech, etc.

Maybe (just maybe) it might be worthwhile if they ensure Voip calls get priority on their network to the destination. Of course that won't work once it breaks out over the internet (no QoS there).

Problem is, line rental is on the way out, fixed line installs are declining, too expensive for what you get. Maybe this is the first step in Eircom ditching POTS:rolleyes:

Cabaal

did they not launch this already?

watty

gerryo wrote:

Maybe (just maybe) it might be worthwhile if they ensure Voip calls get priority on their network to the destination. Of course that won't work once it breaks out over the internet (no QoS there).

But if you make a phone number based call on Eircom, Digiweb, NTL, Smart or Maket using their own VOIP, it never goes on the internet. If it is VOIP to VOIP it only uses their own network for their own numbers or else goes via a PSTN gateway.

eircom can offer a QOS on dsl based VOIP to phone numbers that no ISP on bitstream wholesale can offer.

Calls between numbers of different VOIP operators go via PSTN/Gateway. Only sip URL calls go between VOIP operators on the internet. But 3rd party VOIP operators (Where VOIP provider and physical network are different providers) always entirely use the Internet for all calls.

The secure VOIP systems use a private subnet for the ATA, so those systems can't at all do peer to peer SIP URL and only work via phone numbers and only connect to other operators via PSTN gateway.

gerryo

watty wrote:

eircom can offer a QOS on dsl based VOIP to phone numbers that no ISP on bitstream wholesale can offer..

This is their big advantage over other (non copper owning VoIP suppliers), Eircom can ensure the VoIP call gets shortest uncongested path and maybe breaks out at the nearest exchange to the called number. Will take a bit of network planning to do that properly, but is very workable.

watty wrote:

The secure VOIP systems use a private subnet for the ATA, so those systems can't at all do peer to peer SIP URL and only work via phone numbers and only connect to other operators via PSTN gateway.

That's a bit of a let down, where's the value in an Eircom SIP phone which uses the PSTN to call another (non Eircom) users SIP phone. It might be better quality (maybe, can't be sure) but it means there are PSTN charges, unless it's a local number & these are free on the price-plan in use by the caller.

Most of the non network (as in circuit switched) owning VoIP provides don't charge for a 076 call, Eircom should do the same.

I can see only one advantage to Eircom customers so far, the possability of several simultaneous calls per residence or business, or the potential for individual numbers for each dweller in the residence.

watty

I don't think you quite understand SIP.

A call using a real number to a different provider, (VOIP or not) ALWAYS leaves IP via a Gateway to PSTN no mater who the VOIP provider is.

Only SIP URL calls (i.e. IP address, not phone number) are Internet Peer to Peer and calls using a real number within the same providers number space.

So calls to people on same VOIP supplier, no matter what their Geographic location or Number, even when a number dialled call, (not SIP URL IP) are linked up as VOIP peer to Peer. If the VOIP operator has their own network, it is not actually on the Internet, just on the backhaul and local connection. It depends on the type of "local connection" the subscriber has as to how QOS may be implemented. Some like rip wave or WiFi have no standard QOS method. Cable & Metro have built in QOS for VOIP. On DSL it depends on DSLAM and operator (Not possible on resold Bitstream).

If the operator is 3rd party like Blueface, then it's always on the real Internet and no QOS management exists.

VOIP calls by any operator would not "break out" at the local exchange, A VOIP <--> PSTN gateway is used, it does not connect to analogue circuits. It converts to 64k PCM per call using the network equivalent of ISDN. On a 21CN network presumably it remains as IP traffic but is converted 64K PCM (G711).

SIP does not just use 076. ANY number can be on SIP. Also generally only calls (any number) to same operator are free, some operators may offer ALL 076 free, but they substantially lose money on all the 076 numbers that arn't theirs as they have to go via PSTN gateway. Different VOIP providers with 076 may not be using the same codec.
SIP to SIP calls without a Gateway only work if both are using same codec. Some SIP proxies may do codec conversion, but many VOIP networks only convert codec at PSTN gateway.

Some 076 numbers are also with providers than may use private IP ranges for QOS management and securitry, these are impossible to route peer to peer SIP to a different provider without some kind of equivalent to NAT for SIP (SIP is not NAT friendly), so 076 to 076 between operators use PSTN gateway for this reason.

Basically the main PSTN backbone is eircom, so essentially all VOIP <--> PSTN gateway operators have to pay eircom for every outgoing call. Also if a call terminates at YOUR gateway, you get to charge the caller's network.

So 061, 021, 01 to 076 or whatever call is free on Digiweb, but only if the destination is a Digiweb number. If it is Blueface or Smart 076, even if your Digiweb number is 076 you get charged.

Digiweb SIP uses 076 and ALL the Irish Geographic numbers and 088. I imagine NTL and Eircom can use all the geo numbers and 076 as well.

Also you can port any existing number to Blueface, Digiweb, NTL, Magnet or Smart VOIP.

eircom's move will hit 3rd party VOIP such as resold Smart 076 or Blueface, it's not competitive on quality with ISP's own VOIP. Obviously eircom can undercut unless the regulator stops them as they only pay themselves for PSTN interconnect. They only payout if calls terminate on non-eircom numbers. It can also work with better QOS in theory than an ISP's own VOIP on a resold Bitstream. This is why the cost and difficulty of LLU is an issue. Bitstream and LLU has almost no profit for eircom's ISP competition and Bitstream because it is a generic service offered by eircom resold by the other ISPs, can only have a chance of proper QOS from eircom for VOIP.

Remember ordinary geographic numbers are no longer eircom only.

Solair

It's a bit like the way unilever sells multiple brands of the same washing powder .. persil, surf and umpteen add on options. It exploits niche markets yet maintains absolute dominance. i.e. old fashioned market segmentation.

If eircom think there's a threat from the likes of blueface, they'll try and keep people with eircom products even if they are duplicating their own PSTN infrastructure.

eircom is migrating to an 'NGN' next generation network, which basically means an end to digital circuit switching and a move over to full IP networks and Multi Protocol Label Switching (MPLS) which ensures that time critical info i.e. streaming voice calls, get priority.

So, at that stage, your local exchange will be providing your phone line using VoIP technology anyway. It's quite possible that some already are, I have no idea how far along eircom are with their migration, but they're certainly pushing it as it means cost savings through convergence of varoius seperate networks.

It doesn't necessarily mean entirely new local exchanges (as is the case for BT). eircom's local switch manufacturers have fairly sensible upgrade paths to NGN/IP networks. So, basically the same voice switch can be modified to function as part of an IP-based network.

The end result will simply be your voice calls actually routing over IP, a move away from ISDN services, and faster DSL type broadband services.

In the UK, BT's hyped its 21CN (21st Century Network) in a huge way. It's a bit of a joke really, as there are already NGNs in full service. This includes the likes of Smart Telecom and Magnet where they've unbundled service already rolled out. Other network operators are just upgrading quietly.

I'd say ultimately you might get to a situation where new broadband subscribers will simply get a DSL line without analogue voice services on it at all. Voice being provided using VoIP and your phone connected to a router, or a full voip digital phone. Multi Protocol Label Switching (MPLS) will know that your voice traffic stays on the eircom IP network while your internet traffic will go off into cyberspace.

End result: perfect voice service / private data service and normal internet access.

VoIP only gets unreliable when it has to run over the general internet.

A lot of the other telcos are freaking out about eircom's plans as there's been no element of consultation with anyone and they're not sure how other licenced operators are going to access their end users as this stuff will ultimately mean a total reorganisation of how traffic is handled across the network and how voice providers interconnect with eircom.

All that being said, the major problem in Ireland's not the core network it's the local copper cables i.e. not enough of them (shared multiplexed lines that can't support data at all) or they're too long / not good quality.

So all this stuff may ultimately mean nothing to the end user unless eircom fix the local loops!

The key issues is this:

Skype, Blueface, Vondage etc are using VoIP technology to bypass the incumbant's network and provide calls cheaply.
Incumbants like eircom in Ireland or BT in the UK will use VoIP as a cheaper and more efficient way of providing their voice services on their network. It has everything to do with simplyfing their network operations and reducing costs and nothing to do with providing cheap calls over the internet.

I'm sure that the term VoIP will be thrown about by eircom, BT, AT&T and plenty of others as the new marketing buzz word and end users will simply be confused

gerryo

watty wrote:

I don't think you quite understand SIP.

No no, I do know SIP, it's just I assumed the Irish residential version worked like a big multi-site corporate environment, where all the Irish VoIP suppliers "peer" (not the correct term, but near enough) with each either, so when an 076 number is dialed by (say) a Freespeech user, the Freespeech proxy server has a list of other Irish VoiP providers servers (like a DNS lookup, but not using DNS) it checks to see who has the dialed number registered, & if it gets a response from (say) a Blueface proxy, it routes the call directly to the destination, i.e, over the internet, ISP to ISP

(I'm omitting all the messaging between the proxies for clarity)

I made this assumption because most Irish Voip providers say there is no charge for an 076 call, so I assumed (wrongly, it seems) these calls were peer to peer over whatever network seperated the calling/called party, i.e, the Internet.

I also assumed there was some central authority who controls Irish 076 numbers, since these are designated VoIP. I also assumed other countries VoIP providers also were able to route calls directly to Irish VoIP users by these foreign users just dialing 35376xxxxxx, in otherwords, your VoIP provider acted as a proxy for internet VoIP calls.

Too many assumption in there:(

watty

They many really mean just other numbers on their network. I know a Blueface 076 CAN NOT do Peer to Peer SIP with a Metro 076. It is definately via PSTN (which is not analogue though).

The big list of phone numbers tells you if the number has an IP on your own SIP server (then it uses peer to peer UDP for the audio) or what operator to route the number to. Then it goes via PSTN gateway.

Big VOIP operators can offer local call rates in a foreign country at lowest price by having a gateway in that country or deal with a VOIP operator in that country.

They still have to pay for INEX traffic. Despite what people may think the Internet is not free. Caps & fair use exist because all the operators pay for traffic, not users or time.

matrim

Nothing like this happens, and is unlikely to happen (at least anytime soon). To do this you would have to do interoperability tests between all providers and as it's likely that some of the providers use different SIP platforms, I can almost guarantee you that there will be some issues between them.

You would also have to add a way for the providers to know who a number belonged to, or else have a user dial the full SIP URL of the number they wanted to dial.

gerryo wrote:

No no, I do know SIP, it's just I assumed the Irish residential version worked like a big multi-site corporate environment, where all the Irish VoIP suppliers "peer" (not the correct term, but near enough) with each either, so when an 076 number is dialed by (say) a Freespeech user, the Freespeech proxy server has a list of other Irish VoiP providers servers (like a DNS lookup, but not using DNS) it checks to see who has the dialed number registered, & if it gets a response from (say) a Blueface proxy, it routes the call directly to the destination, i.e, over the internet, ISP to ISP

(I'm omitting all the messaging between the proxies for clarity)

I made this assumption because most Irish Voip providers say there is no charge for an 076 call, so I assumed (wrongly, it seems) these calls were peer to peer over whatever network seperated the calling/called party, i.e, the Internet.

I also assumed there was some central authority who controls Irish 076 numbers, since these are designated VoIP. I also assumed other countries VoIP providers also were able to route calls directly to Irish VoIP users by these foreign users just dialing 35376xxxxxx, in otherwords, your VoIP provider acted as a proxy for internet VoIP calls.

Too many assumption in there:(

Sponge Bob

And Not forgetting eircom want to set the interoperability standard with NGN IP interconnects over the next 2 years.

It will be whatever the others have NOT invested in and of course the lumpen and useless Comreg will do whatever eircom wants instead of providing any form of lead or standards based framework

gerryo

watty wrote:

They many really mean just other numbers on their network. I know a Blueface 076 CAN NOT do Peer to Peer SIP with a Metro 076. It is definately via PSTN (which is not analogue though).

That's very disappointing, & it's not made clear on some VoIP providers sites.
I suppose it does not really matter to users, but it seems a cop-out still using the PSTN for VoIP calls. There may be a slight reduction in voice quality due to transcoding which could be avoided by having a direct media path.

watty wrote:

They still have to pay for INEX traffic. Despite what people may think the Internet is not free. Caps & fair use exist because all the operators pay for traffic, not users or time.

True!, I guess VoIP operators factor this into budgets when they offer services & set prices.

Blaster99

watty wrote:

But if you make a phone number based call on Eircom, Digiweb, NTL, Smart or Maket using their own VOIP, it never goes on the internet. If it is VOIP to VOIP it only uses their own network for their own numbers or else goes via a PSTN gateway.

That's not correct. A VoIP provider uses the internet for the international bit of the call and routes through someone who connects to PSTN locally at local call rates. How else do you think they can do calls for 1c a minute type of rates?

I'm assuming CPS providers like Telestunt do much the same thing behind the scenes.

watty

Actually exiting to Digital PSTN or NGN /21GN rather than inter operator VOIP improves the quality as there is no way for a SIP provider to maintain QOS when it leaves their own network.

Only SIP URL calls are VOIP all the way outside your own operators numbers. Some VOIP providers ATAs don't have any facility for SIP URL. Some you can actually type a IP number instead of a phone number for a known SIP URL by a specific * # etc prefix.
Only on a PC /Laptop softphone is a SIP url available, and in that case peer to peer Skype may be a better choice with its ability to use any skype client as a proxy for both parties (If firewall rules prevent peer to peer UDP) and it has a funky adaptive codec.

watty

Blaster99 wrote:

That's not correct. A VoIP provider uses the internet for the international bit of the call and routes through someone who connects to PSTN locally at local call rates. How else do you think they can do calls for 1c a minute type of rates?

I'm assuming CPS providers like Telestunt do much the same thing behind the scenes.

Only if they have an arrangement with a suitable operator. Not all VOIP operators do. I explained that earlier. It depends on what kind of arrangement they have as to how much the quality suffers.

Rather_b_diving

Here's a couple of facts

Teleglobe is one of the largest voice carriers in the world and MOST of their interconnects are over the Internet!!

VoIP to VoIP peering is up and running - Xconnect allows voip operators from all over the world to exchange numbers ranges and allow interoperator trafic for FREE - you simply pay Xconnect for the priviledge :-P

INEX VoIP peering is in existence but you need to do bi lateral agreements between operators - digiweb to smart could be setup to allow interoperator calls for free but no transit - or transit at a charge

Most professional voIP operators using SIP will use a session border controller or proxy to protect their networks - proxies can transcode and SBCs can/ should negotiate codecs suitable to both ends so you negotiate codec prior to establishing the call. A worst case maybe G.711A law which most european devices will accept.

eircom are now launching VoIP as they see margins being nibbled away by the other voip operators - they won't offer cheaper calls on PSTN as it would erode too much margin and they need money to pay the debt.

Finally anyone who wants to know more about voip in the near future should check out ETSI TISPAN initiative which aims to assimilate all modes of voice traffic into IP (or NGN for the marketeers)

watty

A SIP proxy is very expensive compared with a SIP server as it carries the voice traffic. A SIP server simply sets up the call and then it is peer to peer. The SIP proxy is only useful for SIP to SIP calls outside your own network. Most calls outside a network (if you are not eircom) are still to real PSTN numbers. So you need a Gateway. That will also handle SIP destinations on a different network.

Some operators might not expend the money on an SIP proxy for the tiny % of calls that could be SIP to SIP to an operator that they have an agreement with.

Of course 3rd party VOIP operators basically almost have to have a SIP proxy (i.e. Blueface) as they have no network of their own.

Cable operators / incumbents may not bother with a SIP proxy till a higher percentage of the destination calls are really SIP numbers.

People Like UPC, Smart, Magent, Digiweb effectively are NGN / 21CN as they never had a legacy ISDN/Analogue voice network. They don't need to worry about "convergence". They are already IP only

gerryo

watty wrote:

A SIP proxy is very expensive compared with a SIP server as it carries the voice traffic.

Not sure what you mean here, are you talking about a media proxy?Or, do you mean a redirect server, which redirects the caller to the destination without proxying any SIP messages. These are able to handle higher traffic than a SIP proxy for equivalent CPU power.

watty wrote:

A SIP server simply sets up the call and then it is peer to peer.

Yes, the voice is direct from peer to peer, but if the sip proxy is using "record route", it stays in the loop. Sometimes used for billing purposes e.g, when the proxy is "guarding" a PSTN gateway & calls to the PSTN are charged to SIP users, record route means the proxy knows when the call terminates & the messages can be used for CDR & billing.

watty wrote:

The SIP proxy is only useful for SIP to SIP calls outside your own network.

Yes, but it can be handy also if there are many users behind a NAT router, all with the same IP address, the clients can register to the proxy & it will be able to route incoming calls to the correct destination.
You can do this without a SIP proxy too, but SIP proxies are able to reroute calls to other destinations, forward to voice mail, etc. Very handy & lots of good free ones out there, especially for Linux.

Rather_b_diving

Any VoIP service provider must use a session border controller or SIP proxy they need to dothis to get around NAT issues but also to protect their core VoIP infrastructure. You also need to have an interconnect to eircom if you want phone numbers to work in ireland so either you have a gateway and PSTN links with eircom yourself or you are borrowing numbers from a 3rd party who has links to eircom so no way around that - check number ranges against comreg and you can see who has an affinity with whom.

There is a bit of work done on the VoIP interconnect with INEX but its upto people to agree to pass calls at zero cost.

Finally eircom would probably have some obligation to offer VoIP calls if they match or below the wholesale rate for PSTN calls as they use this infrastructure themselves however there is lots of room on the International side to reduce their markup to maybe only 600% down from 2000% on some destinations and still make loads of money. If they go below the wholesale rate there'll be lots of OAOs asking eircom for a VoIP interconnect

watty

Ah. but again, some non-3rdparty VOIP providers don't work with NAT at all. Their ATAs live on a separate private subnet on the WAN side of the modem.

Unfortunately SIP was not designed with NAT in mind and Proxies, STUN, TURN and ICE are all "patches" to fix this. Even then some SIP clients can't traverse Firewall/NAT as they don't even have STUN support (E70 phone).