You say you're using PAP2. Do they supply this or you have to buy your own?

What do you mean when it is not good enough for conference call? You hear echo? noise? or just unable to make conference call?

What's their rates for calling POTs line to Singapore?

I see. But, if this is softphone only, that's little difference from the existing popular Skype services, perhaps Skype even offer better quality.

I think Skype Out rates is cheaper than TmNET rates even with the currency conversion. Isn't it?

I am not sure how it sounded like over TmNET but Skype voice quality is very good on almost every Internet broadband accessible places (eg. StarBucks hotspots, hotels, SG, HK, TW and US locations) that I have use.

Have you try TMnet VoIP softphones with these USB phones?

I have bought one of these generic cheap USB phone for my laptop when travelling (no need to use the PC headsets) and they works great with Skype.

This post has been edited by netfan: May 10 2006, 06:25 AM

Great find! Looks like IpTelNow allows you to BYO device. Nice!

Have you try calling another IPTelNow user in Malaysia (either on Jaring or TMnet) or the PSTN line in Malaysia? How's the voice quality like?

I have a PAP2 too that I use with US based FWD provider, which allows me to call another FWD for free but unable to call landlines at the moment. I may want to try check out this IPTelNow too since their call out rates are very reasonable. :-)

This post has been edited by netfan: May 13 2006, 07:35 AM

Skype now offers free calls to any landlines in US and Canada for free until the end of year. So, no need for use up any Skype Out credits for calls in US and Canada.

This post has been edited by netfan: May 20 2006, 12:11 AM

Actually not 'web pages access'. You meant the softphone, which is really a small app on your computer.

Nowadays, it is more convenient to get a TA (Telephone Adapter) like the linksys PAP2 or SP3K (Sipura 3000), which has FXO port to allow direct PSTN connection. This way, you don't have to need a PC to make telephone calls.

Xybirium,

Thanks for posting some interesting news tidbits on VoIP providers.

The news clip about VoxAlot is interesting but in my opinion, their procedure for web-call back feature is not really ground-breaking or can cause big concerns for regular POTs providers for potentially losing revenue over their traditional phone calls. For PSTN-to-PSTN, you have to select two VOIP providers that can give you the best rate for untimed calls. Why the hassle? You still required to have a PC connected to the Internet, and if you have that, why bother, just use a softphone to make the call directly.

Besides, most VOIP already provided hefty discounted rate for IP network to PSTN network. Unless one makes a lot calls over the same number between two providers, web-call back option is not really attractive at all. Besides, their claims of "free" is misleading because of this disclaimer at the bottom says "This service is free for all VoXaLot members during our initial trial period. Once the trial completes, it will be available to Premium VoXaLot members."

Today, a lot (unfortunately, not in Malaysia) of traditional phone service carriers are already embracing VoIP as additional call services within and outside their service areas with their own ATAs (or BYO) and using their regular data network in their core backbone to transport voice calls over to the Internet gateway.

The fears that the incumbent carriers to ban VoIP is really unfounded. What we will see is the incumbent carriers won't ban VoIP but participate in this new business model with their choice of VoIP gear and market it directly to their customers plus value-add it with QoS over their network (meaning their infrastructure). The other VoIP providers that doesn't have any infrastructure may have tougher time to sell later because they can't provide the level of QoS as compare to the incumbent who can.

For example in Malaysia:
TMnet can sell their VoIP with better service quality than 'Ah Beng' VoIP service shop. Why? Because TMnet owns the infrastructure and they can control it.

This post has been edited by netfan: May 23 2006, 12:49 AM

Hmm.. you mean like IP-based Contact Centers (aka Call Centers)?

Poor voice quality can be caused by anything on the network. It could be due to telco network congestion or it could caused by end-user equipment too.

If one has a converged network, and if there's problem, unfortunately, the blame always seems to hit the carrier first.

In order to provide good service quality (for both their data and voice network), both
the corp/enterprise and telco have to work together to build and put together a troubleshooting plan for repairs.

Most telcos should be able offer some form of QoS with SLA to differenciate packet-voice and raw data traffic within the boundary of its network (and also to isolate the troubleshooting domains). But, if the end-users POTS lines are provided by another carrier (like when it bridges off to a mobile phone carrier or another central office that has older equipment), it may be difficult to give service guarantee end to end.

This post has been edited by netfan: May 24 2006, 12:55 AM

I agree. Sometimes, the marketing dept of seems to overhype a product that they're selling. You have to really read the fine print with the magnify glass or have a lawyer to examine it.

Most telcos/carriers are only able to provide SLA for PE to PE within their cloud.

However, as I have found in a recent a RFP briefing with one carrier in US, they're are able to provide CE to CE SLAs and over another carrier networks with an established (private) NNI as well. Somehow, they're have stitched together an extended MPLS network over multi-carriers between two cities approx. 2400 km apart. Come to think of this, this can be difficult if both carriers' class of service model doesn't fit.

The question comes to mind is how does one give QoS priorization over another enterprise voice traffic if both are competitors (and futhermore over multi-carriers)?
eg. who do say my voice traffic is more important than the others? Huh?

Ihsan:
Btw, do you know if any NNI for MPLS is being done in Malaysia? For example: TMnet working with Jaring or maybe Maxis working with Digi on their IP network?

This post has been edited by netfan: May 25 2006, 07:48 AM

Sorry, can't reveal specific but I am working in research and educational network (not in Malaysia) with direct dealings with a lot of Tier1/Tier2 ISPs. However, I still considered myself a newbie in VoIP. Voice is not my specialty either. I am router and switches guy.

In the past, voice is solely the phone technician domain. For example, if the phones doesn't work, just call the phone tech at local telco, he'll come and to fix those bad copper lines or replace your phones.

With VoIP, if the calls have problem, the IP phones or the IP-PBX doesn't work, now, the network sys. admin or IT guy has to get involved. Heck, the noisy user will ask, hey did you bring down my network? Did you prioritize my traffic? blah-blah..
Somehow the network always get blamed.

Another thing, 'Triple Plays' ads hype are begining play on those CEO's mind. Sooner or later, we'll see tech support calls like 'TV doesn't work' - call your network administrator (whatever happen to call the local TV repair man?).

Hmmm, come to think of this, I don't think I like where it is heading.

This post has been edited by netfan: May 25 2006, 08:23 AM

Not a problem at all.

VoIP is a not a new thing for those already familiar with the video conferencing (using H.323) environment. It is a simple transition over to packetized voice network. The data network guy now has to learn and understand the old traditional circuit-switching PSTN to make it all work together.

With so many improvement has been done with SIP protocol (for call setup, directory, etc) today, many manufacturers making their ATAs, IP phones, IP-PBX, all have SIP capabilities, making it fairly easy to interface with one another with different vendors.

Having the global IP network (ie.Internet) all stitched all system together, it only make sense to use this network to transport voice as well (yeah, we all know it is best effort esp for those need to call overseas).

There are many concerns about VoIP, particularly in Security and Emergency Services.

Security: The IT network guys know this. The malwares/viruses that spread to your PC, this can happen to your IP-phones,IP-Based PBX. Your VoIP network is subject to same DDoS attacks that plague the IP network. Without proper planning, the voice network will go down with it. Other disaster issue is: Power outage. Traditional phone can still hear dial-tone. The IP-phones is dead if you don't implement PoE (Power over Ethernet) with PSTN trunk line backup.

Emergency Services: These are good questions to ask those newbie VoIP providers.
What does one (say a regular user) expect, when he/she picks up the phone and dial '999' in an emergency situation? Does emergency service (eg.police) know where you're calling from? Is your VoIP provider aware they're responsible to setup a call service to route your calls with all your info to the emergency service provider? Those are important things to consider before we migrate away from our traditional PSTN service.

This post has been edited by netfan: May 27 2006, 12:42 AM

Try setting up with free FWD or VoIPStunt accounts and call yourself using Phone to PC. If this works with no problem, then it must be your VOIP provider.

Anyway, if you're paying your VoIP service with the supported ATA device, your provider should be able help diagnose it as well. PM me if needing more help.

This post has been edited by netfan: May 27 2006, 12:42 AM

thenobodies:

Your examples of "autodialers" use is mainly for telemarketing business, isn't it?

The voip for business basically may save you on the long distance toll-call charges but telco providers are smarter than this. VoIP for business rate is not the same as the residential service rate. Their rates are most likely to be higher.

Whether it is VoIP or Traditional Circuit Switching PSTN, you still need that "autodialer" mechanism to make that volume calling behind the scene. Making it to work may cost you more.

Be warned, some businesses have actually lost more customers after they have moved to VoIP-PSTN for calling because of poor quality, loss calls and frustating call agents. This is not because of the technology, it is because of poor implementation.

The low cost best effort type of residential broadband type of connection is probably not good enough to provide quality and stable network for running a business. You may have to go for dedicated leased line.

This post has been edited by netfan: May 31 2006, 06:30 AM

Here's something to share: VoIP in Osceola School, Florida, USA

A video clip of a success story of VoIP implementation in a US public school by Cisco Systems and a local solution provider - Data Dimension.

Hmm..I wonder if there's any public school in Malaysia has VoIP?

This post has been edited by netfan: May 31 2006, 06:18 AM

Not sure about Malaysia but when I was working in Singapore, I worked on a project to connect a few technical colleges and universities (like NUS, Nanyang, Raffles) with Polycom units. The video conferencing units at the time were using H.323 with H.263 for video quality. They were using it for distance learning and remote meetings. Also, their hospitals were experimenting with tele-health. The Singapore govt has put up a very good infrastructure, all at 10mbps or higher. Nowadays, there are firmwares available in newer video equipement equipment allows them to use SIP and the latest H.264 (mpeg4) video quality .

I went up to Malaysia, wanted to demo a unit in a school in Johor Baru. Unfortunately, that school only has a 128k leased line from Jaring. Video conferencing requires at minimum 384k (bi-directional) for decent quality.

Even with broadband service (like TmNET or Jaring), without synchronous communication capability (meaning same speed both upload and download at minimum 384k), the video looks crappy. It is necessary to get a dedicated leased line (preferable T1/E1 or better) before moving to video service.

Those organisations with video conferencing capability, deploying a voice-ip system is fairly easy. Their video network already has QoS, MCU, Gatekeeper, supporting packetized voice is just another application.

This post has been edited by netfan: May 31 2006, 08:36 PM

Hey rattan, what do you mean by Gizmo Project are con men ?
It seems that their VoIP service is still available.

This post has been edited by netfan: Aug 15 2006, 08:48 AM

Cruzzmz:

Are those PAP2 unlocked or do they lock it to a single provider only?

Also, are you use PAP2 with free VOIP provider like FWD for in-network free calls?
Just wondering what's the quality like when using Streamyx (maybe with package 512k/256k).

cruzzmz:

Personally, did you experience any drop calls or echo or delay when you make calls overseas using your Streamyx connection (up to 300ms in latency)?

What is the cheapest SIP ATA device (min. 1 line) with reasonable features that you can buy in Malaysia today?

This post has been edited by netfan: Aug 15 2007, 08:11 PM