VoIP FAQ

  A VoIP or broadband phone service can help you save up to 90% of your monthly phone bill. VoIP providers offer many features which are covered in the 
monthly fee, such as voicemail, caller ID, three-way calling, and call waiting. Many providers offer unlimited long-distance with their plans, which saves 
customers the trouble of having to purchase long-distance phone cards. Furthermore, most providers offer unlimited calling to other subscribers of the 
same service, allowing you to talk to those users for as long as you wish at no additional cost.

No, Skype is a free software programme that utilises a peer to peer Voice over IP application, meant for home use and not stable or secure enough for 
business. Skype is an application driven by a windows OS. Criticisms of Skype stands due to its "closed" architecture, and if you have a more powerful 
machine it may take over some of the machine's resources to become a "node" and support the calls of other Skype users. Skype does not use pure SIP and so 
is will not allow users to choose universal providers or equipment.

Yes. Your computer and the VOIP service will share your Internet Connection

SIP is a protocol to make voice and video calls over the Internet.
Behind voice and video, you can also use it for instant messaging, presence information, conferencing and more.
You can easily change your current landline phone provider to a SIP based provider.
Internet telephony is usually cheaper, better quality and comes with lots of add-ons (e.g. ability to make video conferences)
You need to follows:
-search one or more good VOIP provider (link)
-get your username/password and the server address
-get a softphone (for example MizuPhone) or a hard-phone (for example Cisco IP Phone)
-start to make phone calls over the internet
 

Voice over IP is the same as Voice over Internet Protocol, and is better known as VoIP.

VoIP refers to voice calls that are routed over online networks using the Internet Protocol--the IP that serves as the backbone of the Internet and is 
used to ferry e-mails, instant messages and Web pages to millions of PCs or cell phones.

VoIP is the transmission of voice communication through IP packets and, therefore, through the Internet. VoIP can use accelerating hardware to achieve 
this purpose and can also be used in a PC environment. VoIP – Voice over Internet Protocol (also called IP Telephony, Internet telephony, and Digital 
Phone) – is the routing of voice conversations over the Internet or any other IP-based network.

Voice over IP (VoIP) can facilitate tasks and deliver services that might be cumbersome or costly to implement when using traditional PSTN.

    * Incoming phone calls can be automatically routed to your VoIP phone, regardless of where you are connected to the network. Take your VoIP phone with 
you on a trip, and wherever you connect to the Internet, you can receive incoming calls.
    * Free phone numbers for use with VoIP are available in the USA, UK and other countries from organizations such as VoIP User.
    * Call center agents using VoIP phones can work from anywhere with a sufficiently fast and stable Internet connection.
    * Many VoIP packages include PSTN features that most telcos normally charge extra for, or may be unavailable from your local telco, such as 3-way 
calling, call forwarding, automatic redial, and caller ID.
 

In order to use VoIP services you can get away with just your PC, an internet connection, and your speakers/microphone. Using a headset is a much better 
solution to avoid echo. You will need the followings:
-A high-speed internet connection: A typical DSL link, for instance, is enough for eight simultaneous phone calls.
-a VoIP client software for example MizuPhone
-An account with a VoIP service provider.

We recommend that you use VoIP services if you have a high speed Broadband Internet connection since the quality of the calls is only as good as your 
connection speed. Tests of VoIP using dial-up services have produced dropped calls, call lagging, and other call quality issues, but if your provider 
allow high quality codec's such as g.729, than you might not have any issues.

You will be able to make long-distance calls less expensive by removing some of the access charges required for use of the public telephone network. A 
user's physical location also becomes irrelevant. VoIP also can power videoconferencing services, or lead to specialized new applications -- like checking 
voice mail from a Web page or programming call-forwarding so calls from work get routed to voice mail while calls from your sweetheart go to your cell 
phone.

When you are using Public Switched Telephone (PSTN) line, you typically pay for the time you use: The more time you stay on the phone the more you'll pay. 
And you generally don't have the option of talking with more than one person at a time (or you can, but at increased cost).

With VoIP, you can talk all the time with any person you want (the requirement is that the other person has an internet connection), with no regard to 
distance, and you can talk with many people at the same time. At the same time, you can exchange data with people are you talking with, sending images, 
graphs and videos.
 

VoIP is still relatively new in the market place and a stable platform is essential for business continuity, SIP providers can often be based on free 
software which can be unreliable if deployed incorrectly.
 

A STUN (Simple Traversal of User Datagram Protocol [UDP] Through Network Address Translators [NATs]) server allows NAT clients (i.e. computers behind a 
firewall) to setup phone calls to a VOIP provider hosted outside of the local network.

The STUN server allows clients to find out their public address, the type of NAT they are behind and the internet side port associated by the NAT with a 
particular local port. This information is used to set up UDP communication between the client and the VOIP provider and so establish a call. The STUN 
protocol is defined in RFC 3489.

The STUN server is contacted on UDP port 3478, however the server will hint clients to perform tests on alternate IP and port number too (STUN servers 
have two IP addresses). The RFC states that this port and IP are arbitrary.

More information about STUN and VoIP in general can be found in our SIP / VoIP Video tutorials, 'Voip Nuggets'. VoIP Nuggets are short youtube technical 
training tutorials about VoIP. Click here for the latest list of VoIP Nuggets

Stun functionality is seamlessly handled by Mizu VoIP server and clients.

Reasons to choose the PBX offered by MizuTech are that MizuTech has more than 10 years of experience in the field of VoIP communications, the software is, fast, robust and reliable software, written in C / C++. Using the server program as PBX is just one way of using it, it can be set up as call center, SIP server, Media server, etc, so the manufacturer has experience in a broad range of IP Telephony related solutions. Our price management’s goal is to offer our services at affordable prices. The problems of our clients are the most important problems we handle. The MizuTech PBX is as feature-rich as any other system currently on the market, and it is under constant development to keep it’s market share and position.

A SIP server is the main component of an IP PBX, dealing with the setup of all SIP calls in the network. A SIP server is also referred to as a SIP Proxy 
or a Registrar.

Although the SIP server is the most important part of the SIP based phone system, it only handles call setup and call tear down. It does not actually 
transmit or receive any audio. This is done by the media server in RTP.

1xx = informational responses

    * 100 Trying
    * 180 Ringing
    * 181 Call Is Being Forwarded
    * 182 Queued
    * 183 Session Progress

2xx = success responses

    * 200 OK
    * 202 accepted: Used for referrals

3xx = redirection responses

    * 300 Multiple Choices
    * 301 Moved Permanently
    * 302 Moved Temporarily
    * 305 Use Proxy
    * 380 Alternative Service

4xx = request failures

    * 400 Bad Request
    * 401 Unauthorized: Used only by registrars. Proxys should use proxy authorization 407
    * 402 Payment Required (Reserved for future use)
    * 403 Forbidden
    * 404 Not Found: User not found
    * 405 Method Not Allowed
    * 406 Not Acceptable
    * 407 Proxy Authentication Required
    * 408 Request Timeout: Couldn't find the user in time
    * 410 Gone: The user existed once, but is not available here any more.
    * 413 Request Entity Too Large
    * 414 Request-URI Too Long
    * 415 Unsupported Media Type
    * 416 Unsupported URI Scheme
    * 420 Bad Extension: Bad SIP Protocol Extension used, not understood by the server
    * 421 Extension Required
    * 423 Interval Too Brief
    * 480 Temporarily Unavailable
    * 481 Call/Transaction Does Not Exist
    * 482 Loop Detected
    * 483 Too Many Hops
    * 484 Address Incomplete
    * 485 Ambiguous
    * 486 Busy Here
    * 487 Request Terminated
    * 488 Not Acceptable Here
    * 491 Request Pending
    * 493 Undecipherable: Could not decrypt S/MIME body part

5xx = server errors

    * 500 Server Internal Error
    * 501 Not Implemented: The SIP request method is not implemented here
    * 502 Bad Gateway
    * 503 Service Unavailable
    * 504 Server Time-out
    * 505 Version Not Supported: The server does not support this version of the SIP protocol
    * 513 Message Too Large

6xx = global failures

    * 600 Busy Everywhere
    * 603 Decline
    * 604 Does Not Exist Anywhere
    * 606 Not Acceptable
 

VoIP is no different than traditional telecommunications. You can call any PSTN number or mobile number using an IP PBX or Hosted account. An IP PBX has 
the added benefit that SIP calls offer cheaper rates than PSTN minutes and that line rentals are much cheaper.

The procedure for placing a VoIP call will depend on the type of equipment you are using. If you are using the PC to Phone/ PC to PC/Phone to PC system, 
you only need a PC, headset and internet connection to place the call. You simply use the dialing features available in Windows to click on the numbers 
you want to make the call. If you are using an IP phone, then the device comes with a regular telephone handset alphanumeric arrangement. You plug the 
Ethernet plug at the end of the cord of the phone to your PC and then press the appropriate buttons to make the call. If you have the ATA equipment, then 
you need to power it up, then place the phone jack at the end of the device into your regular phone, then dial the number you want.

The ability to make a telephone call directly into an internal extension without having to go through the operator.


DID - Direct Inward Dialing (also called DDI in Europe) is a feature offered by telephone companies for use with their customers' PABX system, whereby the 
telephone company (telco) allocates a range of numbers associated with one or more phone lines.

Its purpose is to allow a company to assign a personal number to each employee, without requiring a separate phone line for each. That way, telephony 
traffic can be split up and managed more easily.

DID requires that you purchase an ISDN or Digital line and ask the telephone company to assign a range of numbers. You then need DID capable equipment at 
your premises which consists of BRI, E1 or T1 cards or gateways.
 

Centrex is a set of specialized business solutions (primarily, but not exclusively, for voice service) where the equipment providing the call control and service logic functions is owned and operated by the service provider and hence is located on the service provider's premises. Since Centrex frees the customer from the costs and responsibilities of major equipment ownership, Centrex can be thought of as an outsourcing solution.
Call control and service logic refer collectively to the functions needed to process a telephone call and offer telephone features. The following are examples of call control and service logic functions:
•    recognizing that a party has gone off hook and that dial tone should be provided
•    interpreting the dialed digits to determine where the call is to terminated
•    determining whether the called party is available, busy, or has call forwarding, and then applying the appropriate treatment (e.g., ringing the phone, applying busy signal, applying a call waiting tone, delivering the call to voicemail, or forwarding the call to another party)
•    recognizing when the called party answers the phone and when either party subsequently hangs up, and recording the appropriate information for billing
In traditional Centrex service (i.e., analog Centrex and ISDN Centrex), call control and service logic reside in a Class 5 switch located in the Central Office. The Class 5 switch is also responsibility for transporting and switching the electrical signals that carry the callers' speech or other information (e.g., faxes). Traditional Centrex service has a number of benefits that are discussed elsewhere on this site.
Packet + Centrex = IP Centrex
In IP telephony, voice conversations can be digitized and packetized for transmission across the network. IP Centrex refers to a number of IP telephony solutions where Centrex service is offered to a customer who transmits its voice calls to the network as packetized streams across a broadband access facility. IP Centrex builds on the traditional benefits of Centrex by combining them with the benefits of IP telephony. One of these IP telephony benefits is increased utilization of access capacity. In IP Centrex, a single broadband access facility is used to carry the packetized voice streams for many simultaneous calls. When calls are not active, more bandwidth is available for high speed data sessions over the LAN, like Internet access. This is a much more efficient use of capacity than traditional Centrex. In analog Centrex, one pair of copper wires is need to serve each analog telephone station, regardless of whether the phone has an active call; one the phone is not engaged in a call, the bandwidth capacity of those wires is unused. An ISDN BRI can support two simultaneous calls (i.e., 128 kbps), but similar to analog lines, an idle BRI's bandwidth capacity cannot be used to increase the corporate LAN's interconnection speed.

It’s a phone system used in businesses that has a built in virtual receptionist, that routes calls to national and international destinations, that can manage messages has power features like call queuing, inbound faxing, integrated conferencing, call recording, advanced call transfers, best route selection etc.. You can use your existing office phones, home phones, mobile phones, VoIP phones, SIP softphone, and the easy-to-use Webphone that is available only from MizuTech.
How does it Work?
Set up as a PBX, the Mizu VoIP server routes incoming calls, instant messages, video calls, faxes and data transfers to employees (in case of telephone calls to their voicemail) or to other colleagues if they are busy. It can route to other companies PBX-s, SIP servers, media servers, and conference servers as well.
 

SIP, short for Session Initiation Protocol is an IP telephony signaling protocol used to establish, modify and terminate VOIP telephone calls. SIP was 
developed by the IETF and published as RFC 3261

SIP describes the communication needed to establish a phone call. The details are then further described in the SDP protocol.

SIP has taken the VOIP world by storm. The protocol resembles the HTTP protocol, is text based, and very open and flexible. It has therefore largely 
replaced the H323 standard.

SIP is the internet standard for real time voice and video communication. SIP (Session Initiation Protocol) was developed by the IETF and published as RFC 
3261.

SIP is an internet protocol for live communications used in setting up and tearing down voice or video calls. It is a signaling protocol used to create, 
modify, and terminate sessions with one or more participants in an IP network. A session can be a straightforward two-way phone call or it can be a 
multi-media conference session with many persons participating. SIP has made possible an array of services that seemed unthinkable just a few years ago: 
internet conferencing, IP telephony, instant messaging, presence, voice and video communication, data collaboration, online gaming, application sharing, 
and much more.

SIP is doing for real-time communications what HTTP did for the web and SMTP for email. It is the main driver in the acceleration of the IP Telephony 
revolution. With SIP Telephony, a viable alternative to traditional PBX has emerged. SIP telephone systems deliver features that enhance users’ mobility 
and productivity, while securing substantial cost-saving advantages. This is making proprietary hardware based PBXs obsolete.
 

In terms of technology or how you dial the number, there’s no difference. Call charges, however, vary from plan to plan. Some VoIP providers offer 
unlimited long-distance in which case the call is free. Even when the call is not free the rates are usually very low. Some VoIP providers charge the same 
way as a traditional wire-line telephone service. Others permit you to call anywhere at a flat rate for a fixed number of minutes.

  Yes, your VoIP provider may permit you to select an area code different from the area in which you live. This means, if you live in Austin and get a New 
York number, you will NOT incur long-distance charges while calling a New York number regardless of geography. It also means that your "local" calls, in 
Austin , will be charged long distance and that your friends in Austin will incur long-distance charges calling your New York Number.
Additionally, most providers offer virtual numbers (sometimes known as “extra numbers”). With this feature you may obtain a number from a different area 
code in addition to your current area code. This would enable you to have relatives living in another area code dial a local number to reach you, saving 
them the cost of long-distance charges.

SMS callback module receives the messages with the information on the telephone number (callerID) the message originated from and then, after having analyzed the command, initiates the appropriate operations.

The H.323 standard from the ITU-T provides a foundation for audio, video, and data communications across IP-based networks, including the Internet. These 
networks dominate today’s corporate desktops and include packet-switched TCP/IP and IPX over Ethernet, Fast Ethernet and Token Ring network technologies. 
Therefore, the H.323 standards are important building blocks for a broad new range of collaborative, LAN-based applications for multimedia communications. 
It includes parts of H.225.0 - RAS, Q.931, H.245 RTP/RTCP and audio/video codecs, such as the audio codecs (G.711, G.723.1, G.728, etc.) and video codecs 
(H.261, H.263) that compress and decompress media streams.

Media streams are transported on RTP/RTCP. RTP carries the actual media and RTCP carries status and control information. The signalling is transported 
reliably over TCP. The following protocols deal with signalling:

RAS manages registration, admission, status.
Q.931 manages call setup and termination.
H.245 negotiates channel usage and capabilities.
H.235 security and authentication.

Most VOIP equipment available today follows the SIP standard. Older VOIP equipment though would follow H 323.

FOIP stands for Fax over IP and refers to the process of sending and receiving faxes via a VOIP network.

Fax over IP works via T38 and requires a T38 capable VOIP gateway as well as a T38 capable fax machine, fax card or fax software. Fax server software that 
can talk 'T38' allows sending and receiving faxes directly via a VOIP gateway and, consequently, does not need any additional fax hardware.

3CX includes a T38 compatible network fax server in its 3CX Phone System for Windows. Faxes are converted to PDF files and forwarded via email. Outbound 
faxes are sent via Microsoft Fax from anywhere in the network. Other fax servers currently in the market require the use of separately licensed and 
expensive Dialogic SoftIP drivers.

 

You may need a softphone / ATA device / an IP Phone / a PC 386 or higher and a headset. If you are setting up a PC to PC VOIP connection you may need the followings:

• PC 386 or higher
• Sound card, full duplex capable
• Network card
• High Speed Internet Connection. It can be Cable, Broadband or DSL.
• Microphone and Speakers or a headset.

A VOIP Phone System / IP PBX system consists of one or more SIP phones / VOIP phones, an IP PBX server and optionally includes a VOIP Gateway. The IP PBX 
server is similar to a proxy server: SIP clients, being either soft phones or hardware based phones, register with the IP PBX server, and when they wish 
to make a call they ask the IP PBX to establish the connection. The IP PBX has a directory of all phones/users and their corresponding SIP address and 
thus is able to connect an internal call or route an external call via either a VOIP gateway or a VOIP service provider.

Yes. Both mizu softphone and server has support to CLIP/CLIR services.

•    Account Codes - track and manage telecom expenses.
•    Anonymous Call Rejection - automatically reject incoming calls from parties who do not deliver their name or telephone number with the call.
•    Automatic Callback/Ring Again - when encountering a busy signal, a caller can dial an activation code and be automatically called back when the called station becomes idle.
•    Automatic Line/Direct Connect ("Hotline") - automatically dials a pre-assigned Centrex station's extension number or external telephone number whenever a user goes off-hook or lifts the handset.
•    Barge In - allows a user to bridge himself or herself on to a existing call (at another Centrex station) to form a three-way conference call. Barge-In is often used in conjunction with Directed Call Pickup.
•    Call Block - automatically reject incoming calls placed from specific telephone numbers.
•    Call Forwarding (Busy, Don’t Answer, Multiple Simultaneous, Variable, Selective) - automatically routes incoming calls to a given extension to another preselected number under a variety of circumstances. Call Forwarding Busy forwards calls when the called extension is busy. Call Forwarding Don't Answer forwards calls when there is no answer after a specified number of rings. Call Forwarding Multiple Simultaneous indicates the number of forwarded calls (originally dialed to the same Centrex extension) that can occur simultaneously. Call Forwarding Variable allows users to forward all calls to their extensions to another number (that they select). There are various call forwarding options that allow differential call forwarding to be applied depending on whether the caller and/or the forwarded number are members of the Centrex group or external lines. In addition, Selective Call Forwarding allows the user to pre-select which calls will forward to a different telephone number, based on the based on the calling party’s telephone number.
•    Call Hold (Hard Hold) - calls can be put on hold by dialing a feature activation code (phone does not need a Hold button). After a call is put on hold, the user may perform some task related to the call (e.g., look up customer information), originate another call, answer another call by using a Call Pickup feature, answer an incoming call with the Call Waiting feature, or return to a previously held call.
•    Call Park - allows user to place call on hold, move to a different location, and then resume the call from any other station in the Call Park group.
•    Call Pickup - The lines (or a portion of the lines) in a Centrex group can be made members of a pickup group. A call ringing on any station in the pickup group can be answered from any other station in the pickup group.
•    Call Restrictions/Station Restrictions - prevents certain types of calls from being made or received by particular stations. For example, phones in public areas can be blocked from originating calls to external numbers to prevent unauthorized users from incurring toll charges. Phones in certain areas may be blocked from receiving external calls to limit employees abilities to receive personal calls. A wide variety of restrictions is available which can address incoming calls, outgoing calls, toll restrictions, code restrictions, and differential treatment for internal and external calls.
•    Call Return - allows user to originate a call to the last party or number that called the user, regardless of whether the user answered the original call or knows the caller's identity.
•    Call Selector - uses a special ringing pattern to alert called user of incoming calls from pre-selected telephone numbers.
•    Call Transfer - transfers an existing call to another party (inside or outside the Centrex group)
•    Call Waiting Originating - When a Centrex user (who is assigned the Call Waiting Originating feature) places a call to another Centrex user whose line is engaged, the calling party will hear ringing (instead of a busy signal) and the called party will hear the Call Waiting tone. If the calling user's line has Call Waiting Originating, the called user's line does not need Call Waiting Terminating in order for that user to receive the Call Waiting tone. Upon hearing the Call Waiting tone, the called party can put the current conversation on hold to answer the incoming call.
•    Call Waiting Terminating - alerts the user to incoming calls when the user's line is engaged on an established call. Upon hearing the Call Waiting tone, the called party can put the current conversation on hold to answer the incoming call. Different Call Waiting tones are available to indicate whether the caller is on an outside line or is part of the Centrex group. Tone Block/Cancel Call Waiting is a related feature that allows a user to disable Call Waiting tones for the duration of call so that they are not interrupted.
•    Caller ID - allows the user to identify the name and telephone number of a calling party before answering an incoming call. Another version of this feature--Caller ID on Call Waiting--allows for the calling name and number to be delivered when the called party is on another call.
•    Calling Number Delivery Blocking - prevents a caller's telephone number and/or name from being divulged to the called party (who might otherwise receive that information if they subscribe to Caller ID).
•    Consultation Hold - calls can be put on hold by depressing the switch-hook or pressing the flash button. After completing a second call, the user is automatically reconnected to the originally held call.
•    Code Restriction - prevents a user from dialing one or more three-digit codes. Code Restriction can be used to reduce per call charges for certain services or restrict access to long distance carriers (other than the company's pre-selected long distance carrier).
•    Dial Call Waiting - allows the user to automatically send a Call Waiting tone to another Centrex user when the called party's line is engaged. If the calling user invokes Dial Call Waiting, the called user's line does not need Call Waiting Terminating in order for that user to receive the Call Waiting tone. Upon hearing the Call Waiting tone, the called party can put the current conversation on hold to answer the incoming call. Dial Call Waiting is activated on a per call basis, so the caller can decide to use it only when the call is important enough to interrupt an ongoing conversation.
•    Directed Call Park - allows user to place call on hold, specify the extension number from which the call will be resume, and subsequently move to that location and resume the call.
•    Directed Call Pickup - allows a call ringing at a Centrex station to be answered at a different station. At the station where the call is to be answered, the user dials a feature code and extension number of the ringing telephone. If the user does not finish dialing prior to someone else answering the call, then the user hears a busy signal (if the Barge-In feature is not assigned) or is bridged onto to the call to form a three-way conference call (if the Barge-In feature is assigned).
•    Distinctive Ringing - uses a special ringing pattern to indicate to the called user whether an incoming call is from outside or from within the Centrex group. If the user also has Call Waiting Terminating, then the Call Waiting tones will also be distinctive based on the origin of the call.
•    Executive Busy Override - allows user to bridge onto a busy line and establish a three-way call. This is similar to Directed Call Pickup with Barge-In, except that Executive Busy Override provides a warning tone prior to bridging the call.
•    Intercom Dialing - allows user to call Centrex extensions by dialing a standard 4-digit code instead of the entire 7-digit telephone number.
•    Hunt Groups - allows calls to be redirected to other predetermined lines when the called line is busy. Hunting allows a number of lines to be grouped into a "pool" so that an incoming call is directed to whichever of the lines is available. There are a number of different hunting options which determine how an available line is selected.
•    Last Number Redial - allows user redial the last number called by dialing an access code or by pressing a single button.
•    Message Waiting Audible - provides the user with an audible notification--a "stutter" dial tone--when messages have been left in the company's voice mail system. Centrex service provides a Simplified Message Desk Interface (SMDI) interface so that the company's (or a third party's) messaging system can activate stutter dial tone on specific lines (and deactivate it after the messages have been retrieved).
•    Message Waiting Lamp - provides the user with a visual indication when messages have been left in the company's voice mail system. The indication may be a flashing lamp on a compatible telephone or on an adjacent visual message waiting device.
•    Music-On-Hold - provides a musical interlude for callers who are waiting on hold.
•    Repeat Dialing - automatically dials the last telephone number the user called, and, if that number is busy, continues to monitor the busy line and establishes the call when the line becomes idle.
•    Speed Dialing - allows the user to call frequently called telephone numbers by dialing an abbreviated speed calling code instead of the entire number.
•    Station Message Detail Recording (SMDR) - allows the corporate telecom manager to receive call detail records on a per-station basis before the monthly telephone bill is even issued. SMDR helps the customer control telephone fraud and abuse, perform accurate cost accounting, and analyze call patterns to identify opportunities for cost reductions.
•    Three-Way Conferencing - allows user to add a third party to an existing conversation forming a three-way conference call.

There are two main sources of echo in telephony networks: acoustic echo and  hybrid echo. Acoustic echo is generated on any phone (IP or otherwise) when 
there is feedback from the speaker to the microphone. This is particularly noticeable on many speaker phones. Hybrid echo is very common in the PSTN 
network, and this most commonly occurs when there is a two-wire to four-wire conversion in the network (for example, where analog is converted into T1 or 
E1).

To combat these types of echo, you should always use a headset with softphone's and turn on the acoustic echo canceller (AEC)

Yes, all our software has built-in call forward capability.
Please be aware that the above solution will be billed as follows:
Based on the scenario that A calls B, then B forwards to C.
B will be specified as the originator of the call and will bear all charges for the call.
A will have a duration of 0 seconds and not be billed.

   
VoIP Hunt Groups allow for inbound calls to be automatically routed to multiple extensions until the call is actually answered.

Hunt Groups may be configured so that phone lines ring sequentially or simultaneously (the first person to pick up takes the call).

They are a useful tool in high volume, rapid response time customer service situations.

RTCP stands for Real Time Transport Protocol and is defined in RFC 3550. RTCP works hand in hand with RTP. RTP does the delivery of the actual data, where 
as RTCP is used to send control packets to participants in a call. The primary function is to provide feedback on the quality of service being provided by 
RTP.
 

The Real-time Transport (RTP) Protocol provides end-to-end network transport functions suitable for applications transmitting real-time data such as 
audio, video or simulation data, over multicast or unicast network services. RTP does not address resource reservation and does not guarantee quality-of-
service for real-time services. The data transport is augmented by a control protocol (RTCP) to allow monitoring of the data delivery in a manner scalable 
to large multicast networks, and to provide minimal control and identification functionality. RTP and RTCP are designed to be independent of the 
underlying transport and network layers. The protocol supports the use of RTP-level transla tors and mixers.

A communications channel between two points, typically referring to large-bandwidth telephone channels between switching centers that handle many 
simultaneous voice and data signals.

No calls can be received on a DIDless service, therefore voicemail is not required.
You can only use the DIDless service for outgoing calls.
 

The implementation of an IP telephone system in a business requires the use of a very specific type of phone: the IP Telephone.

IP Phones are sometimes called VoIP telephones, SIP phones or softphones. These are all the exact same thing and are based on the principle of 
transmission of voice over the internet, or what is better known as VoIP (or voice over internet protocol) technology.
 

Some broadband connections may have less than desirable quality. Where IP packets are lost or delayed at any point in the network between VoIP users, 
there will be a momentary drop-out of voice. This is more noticeable in highly congested networks and/or where there is long distances and/or interworking 
between end points. Technology has improved the reliability and voice quality over time and will continue to improve VoIP performance as time goes on.

A telephone function that transmits the billing number of the incoming call (Caller ID, for example).
 

   

A protocol for IP telephony that enables a caller with a PSTN phone number to locate the destination device and establish a session.

Interactive Voice Response or IVR is a telephone technology that communicates with a caller through configurable voice menus and data in real time. In an 
IVR system, callers are given the choice to select options by pressing digits.

IVR systems can normally handle and service high volumes of phone calls. With an Interactive Voice Response system, businesses can reduce costs and 
improve customers’ experience as Interactive Voice Response systems allow callers to get information they need 24 hours a day without the need of costly 
human agents.

Some IVR applications include telephone banking, flight-scheduling information and tele-voting.

The Mizu VoIP Server has a built-in IVR that is designed to boost the competence of any business by increasing flexibility, simplifying processes 
and reducing costs, at the same time as improving customer satisfaction.
 

A method of setting precedence for a particular type of traffic for QoS.
 

SDP, short for Session Description Protocol, is a format for describing streaming media initialization parameters. It has been published by the IETF as 
RFC 4566. Streaming media is content that is viewed or heard while it is being delivered.
 

An IEEE standard for providing virtual LAN (VLAN) identification and QoS levels. Three bits are used to allow eight priority levels, and 12 bits are used 
to identify up to 4,096 VLANs.