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VOIP: Time to Sell on Features Rather than Price?

If you push past the increasingly annoying Vonage ads that never fail to remind you how lower prices are such a good thing, VoIP’s real power is the features such as voice mail to e-mail, conference calls, call logs and call-forwarding. Yet, the industry is still fixated on selling on price despite the obvious need to improve subscriber numbers and ARPU.

So, it’s interesting that VoipReview.org has come out with a study showing the average number of calling features for VoIP service is increasing while monthly prices remain steady at about $25 a month. This is good new for consumers who are obviously getting more bang for the buck. It should also be good news for VoIP service providers such as Vonage, AT&T 8×8 and SunRocket because it could let them maintain or even boost prices as they offer additional services – akin to what the broadband Internet service providers are doing as they maintain or raises price while offering higher download and upload speeds.

This is the kind of news Vonage needs to improve its financial results and the performance of its stock, which is trading close to the record low ($5.65) today. Vonage will post its fourth-quarter and year-end results on Feb. 15.

Update: Om Malik has a good post on Adobe’s VoIP strategy.

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  • http://www.mhgoldberg.com/blog Mark G

    In a starvation contest, the fat guy always wins. VoIP providers can’t compete if their only advantage is deep discount pricing [http://mhgoldberg.com/blog/2006/08/rebtel.html].

    When POTS pricing is cheaper than some VoIP providers (see my piece about Telehop [http://beatskype.notlong.com/]), VoIP providers need to compete on a different basis.

  • Jeremy

    Mark,

    Personally I think until Rogers/Bell lowers their pricing to the vonage level, I don’t see why vonage’s major selling point is not price. Rogers and Bell are not much cheaper than Bell’s regular land lines. Maybe in the US this is different, but definitely not north of the border.

    Jeremy

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  • http://www.voipbazar.com Imran Malik

    Hi Mark

    Impressive piece of information, let me elaborate more on VoIP. Voice over Internet Protocol has been around since many years. But due to lack of sufficient and affordable bandwidth it was not possible to carry carrier grade voice over Internet Protocol. But since the arrival of low cost internet bandwidth and new speech codecs such as G.729, G.723 which utilizes very low payload to carry carrier class voice it has recently been possible to leverage the true benefits of VoIP. G.723 codec utilizes only 6 Kbps (Kilo Bytes/sec) which is capable of maintaining a constant stream of data between peers and deliver carrier grade voice quality. Lets put this way if you have 8 Mbps internet connection, by using G.723 codec you can run upto 100 telephone lines with crystal clear and carrier grade voice quality. I am also a user of VoIP and have setup a small PBX at home. Since I have discovered VoIP I have never used traditional PSTN service.

    Dear readers, if you have not yet tried VoIP I suggest that you try VoIP technology and I bet you will never want to use the traditional PSTN phone service ever again. VoIP has far more superior features to offer which traditional PSTN sadly cannot offer.

    Also It has recently been possile to carry Video alongwith VoIP by using low payload video codecs. I cannot resist to tell you that by using T.38 passthrough and disabling VAD VoIP can carry FAX transmission, but beaware FAX T.38 passthrough will only work when using wide band protocols such as G.711, a-Law and u-Law.

    By using ATA (Analog Telephone Adapter) which converts VoIP signals into traditional PSTN you can also using Dial-up modems to connect to various dialup services. I wont go in to the details what VoIP can offer, to cut my story short VoIP is a must to have product for every business and individual.

    How VoIP Works

    When we make a VoIP call, a communication channel is established between caller and called party over IP (Internet Protocol) which runs on top of computer data networks. A telephony conversation that takes place over VoIP are converted into binary data packets streams in real time and transmitted over data network, when these data packets arrive at the destination these are again converted into standard telephony conversation. This whole process of voice conversion into data, transmission and data conversion into back voice conversation takes place within less than few milliseconds. That is how a VoIP is call is transmitted over data networks. I hope that now you understand basics of how a VoIP call takes place.

    What are speech codec’s and what role codec plays in VoIP?

    Speech codec play a vital role in VoIP and codec determines the quality and cost of the call. Let me explain you what exactly VoIP codec’s are and how they work. You may have heard about data compression, or probably you have heard about air compressor which compresses a volume of air in enclosed container, VoIP codec’s are no different than a air compressor. Speech codec’s compresses voice into data packets and decompresses it upon arrival at destination. Some VoIP codec’s can compress huge amount of voice while maintaining QoS which means use this type of codec will cost less because it will consume just a fraction of data network. Some codec’s are just not capable of encoding huge amount of voice they simply consume huge amount of data networks bandwidth hence the cost goes up.

    Following is a list of VoIP codec’s along with how much data network bandwidth they consume.

    * AMR Codec
    * BroadVoice Codec 16Kbps narrowband, and 32Kbps wideband
    * GIPS Family – 13.3 Kbps and up
    * GSM – 13 Kbps (full rate), 20ms frame size
    * iLBC – 15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size
    * ITU G.711 – 64 Kbps, sample-based Also known as alaw/ulaw
    * ITU G.722 – 48/56/64 Kbps ADPCM 7Khz audio bandwidth
    * ITU G.722.1 – 24/32 Kbps 7Khz audio bandwidth (based on Polycom’s SIREN codec)
    * ITU G.722.1C – 32 Kbps, a Polycom extension, 14Khz audio bandwidth
    * ITU G.722.2 – 6.6Kbps to 23.85Kbps. Also known as AMR-WB. CELP 7Khz audio bandwidth
    * ITU G.723.1 – 5.3/6.3 Kbps, 30ms frame size
    * ITU G.726 – 16/24/32/40 Kbps
    * ITU G.728 – 16 Kbps
    * ITU G.729 – 8 Kbps, 10ms frame size
    * Speex – 2.15 to 44.2 Kbps
    * LPC10 – 2.5 Kbps
    * DoD CELP – 4.8 Kbps

    Switch to VoIP Today and you will never want to use traditional PSTN ever again.

    Thanks

    -Imran