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G729 is a compressed audio codec with better tolerance for packet loss and jitter than G711. G729 uses 33 Kbps of bandwidth whereas G711 uses 87 Kbps. G729 is compressed but STILL sounds very good in POOR network but G711 sounds better only with good network conditions.

G729 is a compressed audio codec with better tolerance for packet loss and jitter than G711. G729 uses 33 Kbps of bandwidth whereas G711 uses 87 Kbps. G729 is compressed but still sounds very good in poor network but G711 sounds better only with good network conditions.

Signalling is a way of information exchange to ESTABLISHMENT and control of a TELECOMMUNICATION circuit and the management of the network.

Common channel signalling (CAS) uses a dedicated channel for the signalling where Channel Associated Signalling (CAS) conveys signalling information relating to multiple BEARER channels. These bearer channels therefore have their signalling channel in common.

CCS with E1 =30 B channels and 1 D channel
CCS with T1 = 23 B channels and 1 D channel
CAS with E1 = 31 B Channels
CAS with T1 = 24 B Channels
B=Bearer channels responsible for carrying voice signal
D=Data channel responsible for signalling control

Signalling is a way of information exchange to establishment and control of a telecommunication circuit and the management of the network.

Common channel signalling (CAS) uses a dedicated channel for the signalling where Channel Associated Signalling (CAS) conveys signalling information relating to multiple bearer channels. These bearer channels therefore have their signalling channel in common.

CCS with E1 =30 B channels and 1 D channel
CCS with T1 = 23 B channels and 1 D channel
CAS with E1 = 31 B Channels
CAS with T1 = 24 B Channels
B=Bearer channels responsible for carrying voice signal
D=Data channel responsible for signalling control

E1 link contains 32 channels where T1 link contains 24 digital channels and each CHANNEL of both T1/E1 gives you 64 kbps BANDWIDTH.

An E1 link consists of 30B channels and 1D channel and each channel has 64k Bw.

So for E1 – you get 30*64 = 1920kbps.

A T1 link consists of 23B channels and a D channel and each channel has 64k Bw. So for T1 – you get 23*64 = 1472kbps

E1 link contains 32 channels where T1 link contains 24 digital channels and each channel of both T1/E1 gives you 64 kbps bandwidth.

An E1 link consists of 30B channels and 1D channel and each channel has 64k Bw.

So for E1 – you get 30*64 = 1920kbps.

A T1 link consists of 23B channels and a D channel and each channel has 64k Bw. So for T1 – you get 23*64 = 1472kbps

The main difference between E1 and T1 is the data rate. T1 has a data rate of 1.544 mbps and E1 has a data rate of 2.048 mbps.

Other differences between T1 and E1 lies in the number of channels (E1-32 Channel and T1-24 channels) but speed (64 KBPS) remains the same for both links: may be for INTER – CONNECTION between the E1 and T1 LINES. This is interconnected because it is USED for international connectivity purposes.

T1 is used mainly in the United States, Canada, Hong Kong and Japan. E1 is mostly used in Europe.

The main difference between E1 and T1 is the data rate. T1 has a data rate of 1.544 mbps and E1 has a data rate of 2.048 mbps.

Other differences between T1 and E1 lies in the number of channels (E1-32 Channel and T1-24 channels) but speed (64 kbps) remains the same for both links: may be for inter – connection between the E1 and T1 lines. This is interconnected because it is used for international connectivity purposes.

T1 is used mainly in the United States, Canada, Hong Kong and Japan. E1 is mostly used in Europe.

SIP Trunk is a voice call CONNECTION placed over your Internet connection. This VoIP “trunk” (or phone line) connects to a PROVIDER who routes your calls through their gateway and usually has very reasonably lower rates on long distance calls, international calls and in-bound toll free calls.

Additionally, SIP trunks can carry instant messages, multimedia conferences, user presence information, and Enhanced 9-1-1 (E9-1-1) emergency calls.

• Eliminate BRI and PRI subscription fees because SIP trunk connected directly to an Internet telephony service provider.
• Eliminate IP-PSTN gateways (or even your entire PBX)
• Low cost long distance calls, international calls, etc.
• Expansion of lines is dependent on bandwidth, which can easily be INCREASED if needed. It means that with SIP trunking you don’t need to buy lines in blocks of 24 or 32. Instead, you can buy the bandwidth you need in smaller increments.

SIP Trunk is a voice call connection placed over your Internet connection. This VoIP “trunk” (or phone line) connects to a provider who routes your calls through their gateway and usually has very reasonably lower rates on long distance calls, international calls and in-bound toll free calls.

Additionally, SIP trunks can carry instant messages, multimedia conferences, user presence information, and Enhanced 9-1-1 (E9-1-1) emergency calls.

VoIP dial PEERS ROUTE calls to other VoIP systems via IP protocol where POTS dial peers route calls to LEGACY PBX systems via local PORTS which can be ANALOG (like FXS, FXO) or digital (like E1/T).

VoIP dial peers route calls to other VoIP systems via IP protocol where POTS dial peers route calls to legacy PBX systems via local ports which can be analog (like FXS, FXO) or digital (like E1/T).

SLRG (Standard Local Route Groups) ELIMINATES the pairing between the gateway and the Route Pattern, THUS creating a more flexible method of selecting a PSTN gateway.
Because it reduces the number of route patterns that need to be created per country, a huge amount of administrative overhead can be saved, ESPECIALLY for organizations with a LARGE number of sites.

SLRG (Standard Local Route Groups) eliminates the pairing between the gateway and the Route Pattern, thus creating a more flexible method of selecting a PSTN gateway.
Because it reduces the number of route patterns that need to be created per country, a huge amount of administrative overhead can be saved, especially for organizations with a large number of sites.

Often we get this kind of issue with analog phones which create a CRACKLING sound when you talk, but is not common for VoIP provider or INTERNET connection as digital systems don’t generate sounds like this. It’s only POSSIBLE that it could be from the person on the other end who is also on an analog telephone line but if this happens with different PEOPLE then it’s likely your phone. Try to replace the phone.

Often we get this kind of issue with analog phones which create a crackling sound when you talk, but is not common for VoIP provider or internet connection as digital systems don’t generate sounds like this. It’s only possible that it could be from the person on the other end who is also on an analog telephone line but if this happens with different people then it’s likely your phone. Try to replace the phone.

MGCP :  Uses clear text for call controlUses a client-server modelIdeally positioned for service providers (centrally located call agents)Centralized management and control (Dial PLAN, etc.)Enhanced call survivabilityBetter feature interaction with capabilities LIKE Caller-IDSupport of QSIG supplementary services with CCM.

H.323 : Uses Abstract SYNTAX Notation 1 for call control messagesUses a peer-to-peer modelScales WELL in an enterpriseFractional PRI supportCaller-ID support on analog FXOMany more TDM interface types and signallingGateway-resident applications like TCL and VXMLCAC network design with H.323 GatekeepersNo release dependencies between GWs and CCMCall preservation for SRST on PRIsNFAS support

MGCP :  Uses clear text for call controlUses a client-server modelIdeally positioned for service providers (centrally located call agents)Centralized management and control (Dial plan, etc.)Enhanced call survivabilityBetter feature interaction with capabilities like Caller-IDSupport of QSIG supplementary services with CCM.

H.323 : Uses Abstract Syntax Notation 1 for call control messagesUses a peer-to-peer modelScales well in an enterpriseFractional PRI supportCaller-ID support on analog FXOMany more TDM interface types and signallingGateway-resident applications like TCL and VXMLCAC network design with H.323 GatekeepersNo release dependencies between GWs and CCMCall preservation for SRST on PRIsNFAS support

A VoIP gateway works as a bridge between an IP network and the PSTN. It converts analog TELEPHONY SIGNALS to digital.

VoIP gateways include the FOLLOWING features: Call routing, packet processing and control signalling managementVoice and fax compression/decompressionExternal controller INTERFACES, for example to a soft switch, billing system or network MANAGEMENT system

A VoIP gateway works as a bridge between an IP network and the PSTN. It converts analog telephony signals to digital.

VoIP gateways include the following features: Call routing, packet processing and control signalling managementVoice and fax compression/decompressionExternal controller interfaces, for example to a soft switch, billing system or network management system

 PVDM stands for Packet Voice DSP (digital signal processor) MODULE and it enables CISCO INTEGRATED Services Routers to provide high-density voice connectivity, conferencing, and transcoding capabilities in Cisco IP Communications solutions.

 PVDM stands for Packet Voice DSP (digital signal processor) Module and it enables Cisco Integrated Services Routers to provide high-density voice connectivity, conferencing, and transcoding capabilities in Cisco IP Communications solutions.

VoIP service providers offer their solutions over most kinds of BROADBAND INTERNET.
A typical VoIP call only requires about 100 Kbps for best quality. NETWORK latency obviously must be kept LOW for digital phone calls to maintain good sound quality; VoIP over satellite Internet can be problematic, for EXAMPLE.

VoIP service providers offer their solutions over most kinds of broadband Internet.
A typical VoIP call only requires about 100 Kbps for best quality. Network latency obviously must be kept low for digital phone calls to maintain good sound quality; VoIP over satellite Internet can be problematic, for example.

No. Internet phones support NUMBER portability. Those switching from ORDINARY TELEPHONE service to VoIP service can normally keep their same number. Note, however, that VoIP PROVIDERS are normally not the ones RESPONSIBLE for switching your old phone number over to their service. Check with your local phone company as some may not support a number transfer.

No. Internet phones support number portability. Those switching from ordinary telephone service to VoIP service can normally keep their same number. Note, however, that VoIP providers are normally not the ones responsible for switching your old phone number over to their service. Check with your local phone company as some may not support a number transfer.

When the network is functioning well, VoIP sound quality is excellent. So good, in FACT, that some VoIP service PROVIDERS actually INJECT special SOUNDS (called "comfort noise") into the transmission, so that callers don't mistakenly think the connection is DEAD.

When the network is functioning well, VoIP sound quality is excellent. So good, in fact, that some VoIP service providers actually inject special sounds (called "comfort noise") into the transmission, so that callers don't mistakenly think the connection is dead.

Traditional phone LINES can be wiretapped, but this requires PHYSICAL access and INSTALLATION effort. VoIP communications, on the other hand, can be snooped over the Internet electronically. Network ATTACKERS can likewise disrupt your calls by interfering with the FLOW of data packets.Ensure home network security systems are in place to minimize security concerns with VoIP.

Traditional phone lines can be wiretapped, but this requires physical access and installation effort. VoIP communications, on the other hand, can be snooped over the Internet electronically. Network attackers can likewise disrupt your calls by interfering with the flow of data packets.Ensure home network security systems are in place to minimize security concerns with VoIP.

In order to be able to make voice calls by using your own software application you need to connect your system to the telephone network. This can be DONE in three ways:

Option #1: Use a VoIP telephone adapter

A VoIP telephone adapter is a hardware device that can be connected to your Ethernet LAN or to your PC. There are VoIP telephone adapters for GSM lines, for standard ANALOG telephone lines and for ISDN lines. When you connect this hardware to your Ethernet LAN, it will receive an IP address. You need to configure this IP address in your VOIP SDK.

Option #2: Use a SIP ACCOUNT provided by VoIP telephone SERVICE provider

There are many VoIP telephone service providers WORLDWIDE that offer phone service over the Internet. You need to subscribe to their service, and you will receive a SIP account (including an IP address, a username and a password). You need to configure the SIP account details in your VoIP SDK.

Option #3: Use your existing office PBX if it is a VoIP system.

If you already have an IP telephone system, you need to connect your VoIP SDK to that over the LAN. The SDK can log in to the phone system by using a SIP account and it can make telephone calls just like any other

In order to be able to make voice calls by using your own software application you need to connect your system to the telephone network. This can be done in three ways:

Option #1: Use a VoIP telephone adapter

A VoIP telephone adapter is a hardware device that can be connected to your Ethernet LAN or to your PC. There are VoIP telephone adapters for GSM lines, for standard analog telephone lines and for ISDN lines. When you connect this hardware to your Ethernet LAN, it will receive an IP address. You need to configure this IP address in your VOIP SDK.

Option #2: Use a SIP Account provided by VoIP telephone service provider

There are many VoIP telephone service providers worldwide that offer phone service over the Internet. You need to subscribe to their service, and you will receive a SIP account (including an IP address, a username and a password). You need to configure the SIP account details in your VoIP SDK.

Option #3: Use your existing office PBX if it is a VoIP system.

If you already have an IP telephone system, you need to connect your VoIP SDK to that over the LAN. The SDK can log in to the phone system by using a SIP account and it can make telephone calls just like any other

The best way for creating any VoIP application is using a VoIP development KIT. These SDKs are intended to provide a background support for your VoIP project by OFFERING prewritten VoIP components. It is quite effective and comfortable to use these prewritten components, as you can save time and money as well. (During the softphone development that will be described below, Ozeki VoIP SIP SDK has been used for this purpose, that supports all the .NET programming languages, so C# as well.)
For using these toolkits, you have to add your preferred SDK as reference in you IDE. After you have added it, you can REACH all VoIP components that are needed to be ABLE to define the behaviour of such VoIP applications as softphones, call recorders, IVR MENU systems, software-based IP phone systems (PBX), etc.

The best way for creating any VoIP application is using a VoIP development kit. These SDKs are intended to provide a background support for your VoIP project by offering prewritten VoIP components. It is quite effective and comfortable to use these prewritten components, as you can save time and money as well. (During the softphone development that will be described below, Ozeki VoIP SIP SDK has been used for this purpose, that supports all the .NET programming languages, so C# as well.)
For using these toolkits, you have to add your preferred SDK as reference in you IDE. After you have added it, you can reach all VoIP components that are needed to be able to define the behaviour of such VoIP applications as softphones, call recorders, IVR menu systems, software-based IP phone systems (PBX), etc.

The first thing all mean the same thing. Which is using IP (Internet protocol) for voice services. Some voice networks are only packet-switched and have no access outside of their own VoIP network. Most VoIP networks have a GATEWAY that connects to a circuit-switched EXTERNAL network which gives them acces to external calling. One of the gateways responsibilites is to convert G.711 Circuit-switched media (typically a T1 provided by a telco company) to the 7.723 Packet-switched media that will traverse the companies VoIP network. A device called a gatekeeper will then convert the IP address (used by H.323 protocol) to a standard telephone NUMBER (E.164 address) that can be used for external calling.

A converged network is a network that passes both Voice and Data over the same set of devices. Converged networks generally implement QoS (Quality of service) on all actived network devices to ensure the VoIP has priority over standard data because of it's more RIGID demands.

The first thing all mean the same thing. Which is using IP (Internet protocol) for voice services. Some voice networks are only packet-switched and have no access outside of their own VoIP network. Most VoIP networks have a Gateway that connects to a circuit-switched external network which gives them acces to external calling. One of the gateways responsibilites is to convert G.711 Circuit-switched media (typically a T1 provided by a telco company) to the 7.723 Packet-switched media that will traverse the companies VoIP network. A device called a gatekeeper will then convert the IP address (used by H.323 protocol) to a standard telephone number (E.164 address) that can be used for external calling.

A converged network is a network that passes both Voice and Data over the same set of devices. Converged networks generally implement QoS (Quality of service) on all actived network devices to ensure the VoIP has priority over standard data because of it's more rigid demands.

VoIP has been implemented in various ways USING both proprietary protocols and protocols based on open standards. You can see the VoIP protocols below:

• H.323
• MEDIA GATEWAY Control Protocol (MGCP)
• Session Initiation Protocol (SIP)
• H.248 (also known as Media Gateway Control (Megaco))
• Real-time Transport Protocol (RTP)
• Real-time Transport Control Protocol (RTCP)
• Secure Real-time Transport Protocol (SRTP)
• Session DESCRIPTION Protocol (SDP)
• Inter-Asterisk eXchange (IAX)
• Jingle XMPP VoIP extensions
• Skype protocol
• Teamspeak

VoIP has been implemented in various ways using both proprietary protocols and protocols based on open standards. You can see the VoIP protocols below:

The big advantage is VoIP may save you money depending on how much you are currently spending for local and long-distance calls. What you will need to do is get the total cost the phone company is charging and compare it against a VoIP plan that interests you. With most plans, you get free calls within the U.S. and Canada for a low flat rate. International calls usually have very low rates with no connection fees. For both residential customers and BUSINESSES that make a lot of long distance and international calls, the savings can be several hundred dollars a year. 

Another advantage is with the features available with VoIP. Features such as caller ID, CALL waiting, call forwarding, 3 way conferencing and voice mail are usually INCLUDED at no extra cost. With the phone company, these services are usually extra. 

In addition, you can make free phone calls anywhere there is a HIGH speed Internet connection available. That means you can be in another state or even in another country and make calls as if you were back at your HOME or business. You will just need to bring your phone adapter along with you and possibly a phone in case one is not available.

The big advantage is VoIP may save you money depending on how much you are currently spending for local and long-distance calls. What you will need to do is get the total cost the phone company is charging and compare it against a VoIP plan that interests you. With most plans, you get free calls within the U.S. and Canada for a low flat rate. International calls usually have very low rates with no connection fees. For both residential customers and businesses that make a lot of long distance and international calls, the savings can be several hundred dollars a year. 

Another advantage is with the features available with VoIP. Features such as caller ID, call waiting, call forwarding, 3 way conferencing and voice mail are usually included at no extra cost. With the phone company, these services are usually extra. 

In addition, you can make free phone calls anywhere there is a high speed Internet connection available. That means you can be in another state or even in another country and make calls as if you were back at your home or business. You will just need to bring your phone adapter along with you and possibly a phone in case one is not available.

Generally following THINGS are required for voip 

1. BROADBAND connection 
2. voip phone 
3. nexton softswitches 
4. ROUTER 
5. audiocodec 
6. astric SERVER

Generally following things are required for voip 

The first thing all mean the same thing. Which is using IP (Internet protocol) for voice services. Some voice NETWORKS are only packet-switched and have no access outside of their own VoIP network. Most VoIP networks have a Gateway that connects to a circuit-switched external network which gives them acces to external calling. One of the GATEWAYS responsibilites is to convert G.711 Circuit-switched media (typically a T1 provided by a telco company) to the 7.723 Packet-switched media that will traverse the companies VoIP network. A device called a gatekeeper will then convert the IP address (USED by H.323 protocol) to a standard telephone number (E.164 address) that can be used for external calling. 

A converged network is a network that passes both Voice and DATA over the same set of devices. Converged networks generally implement QoS (Quality of service) on all actived network devices to ensure the VoIP has PRIORITY over standard data because of it's more rigid demands.

The first thing all mean the same thing. Which is using IP (Internet protocol) for voice services. Some voice networks are only packet-switched and have no access outside of their own VoIP network. Most VoIP networks have a Gateway that connects to a circuit-switched external network which gives them acces to external calling. One of the gateways responsibilites is to convert G.711 Circuit-switched media (typically a T1 provided by a telco company) to the 7.723 Packet-switched media that will traverse the companies VoIP network. A device called a gatekeeper will then convert the IP address (used by H.323 protocol) to a standard telephone number (E.164 address) that can be used for external calling. 

A converged network is a network that passes both Voice and Data over the same set of devices. Converged networks generally implement QoS (Quality of service) on all actived network devices to ensure the VoIP has priority over standard data because of it's more rigid demands.

If you're considering REPLACING your traditional telephone service with VOIP, there are some possible differences: 

• Some VoIP services don't work during POWER OUTAGES and the service provider may not offer BACKUP power. 
• Not all VoIP services connect directly to emergency services through 9-1-1. 
• VoIP providers may or may not offer directory assistance/white page listings.

If you're considering replacing your traditional telephone service with VoIP, there are some possible differences: 

Early VoIP services often sounded TINNY, delayed or digitized. In recent years, the technology has gotten MUCH better, and Guzman & Co. analyst Pat Comack SAID it won't be long before it achieves the tech industry's vaunted "five 9s" level -- 99.999 percent RELIABILITY. For now, VoIP MAY not be perfect, but cell phones have made people more accepting of less-than-ideal sonic conditions.

Early VoIP services often sounded tinny, delayed or digitized. In recent years, the technology has gotten much better, and Guzman & Co. analyst Pat Comack said it won't be long before it achieves the tech industry's vaunted "five 9s" level -- 99.999 percent reliability. For now, VoIP may not be perfect, but cell phones have made people more accepting of less-than-ideal sonic conditions.

A BROADBAND INTERNET connection. You can use a REGULAR phone, as long as you connect it to an adapter. However, companies such as Cisco SYSTEMS Inc. now make VoIP phones that don't REQUIRE adapters. The adapter or new VoIP phone connects to a broadband modem. Some VoIP providers can link to an entire home's wiring so adapters aren't required at each extension.

A broadband Internet connection. You can use a regular phone, as long as you connect it to an adapter. However, companies such as Cisco Systems Inc. now make VoIP phones that don't require adapters. The adapter or new VoIP phone connects to a broadband modem. Some VoIP providers can link to an entire home's wiring so adapters aren't required at each extension.

The biggest disadvantage is that currently all VoIP providers to not OFFER emergency 911 service in all areas. The FCC has RECENTLY RULED though that this is a MANDATORY requirement that must be COMPLETED by September 2005.

The biggest disadvantage is that currently all VoIP providers to not offer emergency 911 service in all areas. The FCC has recently ruled though that this is a mandatory requirement that must be completed by September 2005.

Most LIKELY in the next couple of years. Right now CELL phone SERVICE providers are developing cell phones that can roam for Wifi VoIP or traditional networks in order to GIVE you the greatest savings dependent upon location. Wifi hotspots, however, do not currently cover as much area as cellular does, so it will take providers some time to WORK this issue out.

Most likely in the next couple of years. Right now cell phone service providers are developing cell phones that can roam for Wifi VoIP or traditional networks in order to give you the greatest savings dependent upon location. Wifi hotspots, however, do not currently cover as much area as cellular does, so it will take providers some time to work this issue out.

Yes, many companies are doing just this. VoIP is a very cost-effective method for companies who want to UPGRADE their old PBX systems and VoIP allows for NEW features that traditional PBX systems simply do not. Also VoIP web CONFERENCING has come of age and also OFFERS cost savings and many new features not available with traditional web conferencing methods.

Yes, many companies are doing just this. VoIP is a very cost-effective method for companies who want to upgrade their old PBX systems and VoIP allows for new features that traditional PBX systems simply do not. Also VoIP web conferencing has come of age and also offers cost savings and many new features not available with traditional web conferencing methods.

You can use VoIP at home currently with a free application such as Skype which lets you TALK from your PC to ANOTHER person who also has Skype on their PC for free ANYWHERE in the world. You can also get a VoIP phone with a VoIP service provider or you can use your regular phone with a VoIP telephone ADAPTER PLUS a service provider.

You can use VoIP at home currently with a free application such as Skype which lets you talk from your PC to another person who also has Skype on their PC for free anywhere in the world. You can also get a VoIP phone with a VoIP service provider or you can use your regular phone with a VoIP telephone adapter plus a service provider.

A VoIP GATEWAY converts the analog voice DATA into digital data PACKETS which are sent over a broadband Internet connection to a VoIP SERVER that forwards the message to the receiving party.

A VoIP gateway converts the analog voice data into digital data packets which are sent over a broadband Internet connection to a VoIP server that forwards the message to the receiving party.

VOIP stands for Voice Over Internet PROTOCOL or Voice Over IP. VoIP is the METHOD by which you can send your voice (analog data) over the Internet (digital data) to an end-user who receives the message.

VoIP stands for Voice Over Internet Protocol or Voice Over IP. VoIP is the method by which you can send your voice (analog data) over the Internet (digital data) to an end-user who receives the message.