The VoIP Warrior – Business VoIP, Hosted IP PBX Solution

Windows based, ISDN and VOIP Ready. Suitable for companies with 1 to 200 employees.
Supports Traditional, USB, SIP Phones, Wireless Headsets, Wifi Phones and Softphones.

The features are comprehensive and includes:

VOIP Ready – Internet VoIP calls over SIP Carriers. ISDN Ready – Connect in existing ISDN lines. VOICEMAIL – by phone, or by email for every extension and service groups. AUTOMATED CALL ATTENDANTS (IVR) Voice/Music Greetings to Callers providing Optional Menus by Touch Tone Dialling. PRESENCE indicators for every extension showing extension on-call, free, away or offline. Outlook Integration – pop up relevant contact details and notes for incoming caller. Call Recording – Automatic or manual instant recording. WAN – Integrate branch offices and benefit from free inter office calls, and improve office communication efficiency. Centralised Phone Book – company wide telephone book showing names, numbers and business information. Fax Server – company wide fax server allowing employees to fax straight from desktop PCs, receive incoming faxes, and notifications of successful/unsuccessful messages. Call Pickup/Conferencing/Divert/hold/Forward Music on Hold – configurable, accepts any .wav file Short Text Messaging – short message any extension on same system or in another branch office. SIP Phones, USB Phones, Softphones and Traditional RJ11 Phones – wide ranging support of hard phones and softphones. Least Cost Routing – automatic routing with calls between VOIP and ISDN carriers to minimize outages and call charges. Day/Night/Holiday Modes – system programmable to offer different messages or call routes depending on day, time and holidays.

For small businesses the SMCDSP-205 can be used with SMC’s TigerVoIP™ IP PBX (SMCPBX10). This allows you to take advantage of free calls between branch offices located around the world as well as reduced rate calls to the traditional telephone network. Increased functionality is added with supplementary services such as Call Waiting, Call Hold, Call Resume, Call Transfer, Call forwarding and 3-way conference calling. You can also centrally manage VoIP user accounts and automatically provision these settings to phones.

The SMCDSP-205 has a stylish and functional design. It includes a LCD display, 10 feature keys, programmable speed dials, a speaker phone and two 10/100Mbps Ethernet ports. The two 10/100Mbps Ethernet ports allow you to connect the phone and a PC using a single network point. Configuring the SMCDSP-205 via the keypad and web based management interface is intuitive making the phone easy to setup and manage.

Allo.com: VoIP PBX and VoIP ATA hardware reviews

Allo.com Nano/Hybrid PBX

• Allo.com Nano/Hybrid PBX
• Full featured VoIP PBX System
• User friendly Graphical User Interface (GUI)
• Compact
• Affordable to home users and small businesses
• Built in voice mail, multi level auto attendants
• Remote extensions
• 3 Way Calling
• For more features visit Nano PBX Features

Allo.com VoIP ATA Phone Systems

• Advanced SIP based VoIP ATA
• Compatible with any VoIP Network
• Global uses
• Web based control panel
• 2 VoIP Lines
• 1 Regular Phone Line
• For more features visit VoIP ATA Phone System

Allo.com VoIP Hardware Ratings:

Features: 5/5

Price: 5/5

Support: 5/5

Delivery: 5/5

Value of Money: 5/5

Please 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 voipbazar.com