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Voice over IP – Wikibooks, open books for an open world

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Contents[edit]

  1. Introduction
  2. Transferring Away from Conventional Telephony
  3. Commonplace Architectures for Subsequent Technology Networks
  4. VoIP Protocals, {Hardware}, and Software program
  5. VoIP Gateways and PABX Integration
  6. Superior VoIP Configuration
  7. ENUM and E146 Expertise
  8. Reference

Introduction[edit]

Voice over IP (or VoIP) is an rising expertise permitting encapsulation of voice telephony companies inside IP (Web Protocol) for transmission over IP networks, such because the Web.

VoIP expertise gives low-cost and versatile voice telephony by using packet-switched networks for voice information, offering a less expensive resolution than conventional circuit-switched telecommunications companies, whereas additionally permitting termination of IP calls nearer to the supposed vacation spot, avoiding long-distance tolls.

With the worldwide prevalence of broadband expertise at residence, curiosity in telecommunication expertise and a transfer away from conventional copper line telephony, there’s loads of trade curiosity in VoIP expertise.

Goal[edit]

This e-book is meant for individuals who possess reasonable to excessive computing expertise. The preliminary sections of the e-book, coping with VoIP historical past and easy VoIP deployment for residence and small enterprise customers, is particularly written to permit these excited about VoIP expertise to get a possibility to attempt it out with out getting too technical. For these extra technically inclined, topics corresponding to VoIP protocol evaluation and superior PBX capabilities are mentioned towards the top of the e-book.

Transferring away from conventional telephony[edit]

There are a lot of compelling causes for people and companies to put money into Voice over IP. For smaller companies or residence customers who continuously make lengthy distance or timed telephone calls, to bigger industrial ventures seeking to cut back prices on site-to-site telephony.

The vary of VoIP plans accessible additionally enable extra flexibility and value management. As a result of VoIP termination doesn’t contain any bodily rental or eat assets when idle, plans can be found on a per-use or pay as you go foundation, on a month-to-month subscription foundation, or in any variety of different configurations.

A profitable VoIP deployment requires good planning and a great understanding of the strengths and weaknesses of IP Telephony. Utilizing the best {hardware}, the best VoIP supplier(s), and a customized dialplan can save vital quantities of cash.

Each the constructive and detrimental features of VoIP expertise are detailed within the following sections.

Advantages of VoIP[edit]

Cheaper long-distance – As a result of VoIP calls will be terminated wherever on the earth, long-distance calls will be made for less expensive charges. For instance, if you’re a European VoIP person who continuously calls US numbers, you should purchase VoIP termination credit score from a US VoIP supplier, which can typically provide close to native name costs from wherever on the earth.

Incoming Calls – For a similar motive as above, VoIP gives the power to acquire incoming phone numbers from wherever around the globe, routed to you over the Web. It is potential to acquire US 1-800 numbers, for instance, it doesn’t matter what your location.

No copper rental – With out the necessity for a circuit out of your residence or enterprise to your phone firm’s alternate, your month-to-month prices can cut back considerably. But, there should be some technique of connectivity from the “Finish Consumer A” who initiates a name onto the Web. Ostensibly, this connection is both a broadband connection (Cable or Telco DSL), or a “dial-up” connection. In each circumstances, and on this sense, clearly a copper rental is required.

Dial-Plan freedom – Utilizing a dial plan, you’ve the power to direct your softphone, {hardware} telephone or VoIP gateway to ship calls to totally different VoIP gateways or by way of land-line, relying on the quantity dialled. You even have the power to set precedence numbers, directing the gateway to drop an current name in favour of the precedence name the place no traces can be found.

Free IP-to-IP Calls – The place each the caller and recipient are utilizing VoIP to put/obtain calls, and the place the proper configuration exists (see the part on ENUM on the finish of this e-book), calls will be positioned completely over the Web with none cost.

VoIP Points[edit]

As with all new expertise, Voice over IP has its share of points which should be thought of fastidiously earlier than implementation.

Emergency and Data companies – There is no such thing as a assure that calls terminated over IP for numbers corresponding to 911 and 411 (in the USA) will likely be appropriately routed. As a result of lack of location-specific termination of Voice over IP calls, it isn’t potential for such companies to find out your location, and it’s potential that the gateway your calls are terminating on doesn’t have entry to such numbers in any respect.

This limitation, and solutions for workarounds, are mentioned later on this e-book.

Bandwidth Points – VoIP protocols will be very delicate to each bandwidth limitations and community latency. Usually, VoIP calls should be positioned by way of a broadband connection. The place there’s heavy bandwidth utilization, excessive community latency, or the gateway is a big distance from the person, name high quality will be closely degraded with the likelihood that there will likely be no VoIP service in any respect.

See the chapter entitled site visitors shaping and bandwidth management later on this e-book for solutions on how this may be managed.

NAT/PAT Points – NAT (Community Handle Translation) and PAT (Port Handle Translation) are used to supply connectivity to machines in lots of conditions corresponding to personal networks which require Web connectivity.

With NAT, a gateway/router system will translate an IP handle from the supply handle supplied by the machine into one other IP handle completely. NAT is commonly confused with PAT; nevertheless NAT gives a 1:1 relationship, eg. 192.168.10.10 will likely be translated to 203.10.10.10.

PAT is a barely totally different translation. With PAT, a number of public IP addresses exist for any variety of personal/translated addresses. In the most typical setup, a house person might have a residential DSL connection and a single public IP handle, nevertheless they could have a number of computer systems on a personal community behind their gateway/router which require Web connectivity.

With PAT, the router/gateway will translate all personal IP addresses into the general public handle supplied, and can preserve observe of which ports established by way of this handle belong to which personal IP addresses, permitting the machines to share the general public IP handle.

VoIP protocols usually have inherent points with NAT and PAT. As a result of prevalence of those gateway units for cable, wi-fi and DSL customers, even customers with single machines might discover that PAT is carried out by default by way of their router.

Please see the part instantly beneath entitled VoIP Protocols for extra data on which protocols specifically are affected by NAT and PAT and find out how to keep away from issues.

Commonplace Architectures for Subsequent Technology Networks[edit]

IMS(IP Multimedia Subsystem)[edit]

IMS is at the moment in improvement and testing phases, and was initially developed by 3GPP as an structure for Cellular Networks.

IMS describes a full structure that makes use of IP applied sciences to move all site visitors (each Voice and Knowledge). It tries to re-use current protocols outlined by IETF and different requirements our bodies as a substitute of re-inventing the wheel. A few of the protocols it reuses are SIP, MEGACO/H.248, SIGTRAN and DIAMETER.

TISPAN[edit]

TISPAN reuses as a lot as potential from the IMS structure and enhances it with the required components to serve mounted/wi-fi networks.

The work developed by TISPAN is then feed again into IMS.

Voice over IP Expertise[edit]

The time period VoIP is commonly used to explain normal IP telephony expertise. Behind this time period lies a spread of various protocols, languages used to speak between a spread of various units and distributors.

These protocols are one of the vital components in selecting a VoIP resolution. It’s typically inconceivable to get totally different units to speak, except they help the identical protocol.

VoIP Protocols[edit]

The next protocols are related to VoIP voice name termination and inter-gateway communication.

H.323 – H.323 is utilized by many industrial distributors for IP telephony, It’s a suite of protocols
which incorporates H.245, Q.931, and so on. This suite takes care of session institution between telephone and swap and is closely based mostly on the Built-in Companies Digital Community (ISDN) signalling protocols. Signalling half can be taken care by this protocol. This suite makes use of RTP, RTCP, RSVP for precise information switch and QOS between telephones.

SIP (Session Initiation Protocol) – SIP is a typical developed by the IETF (Web Engineering Process Drive) to be used in establishing multimedia periods corresponding to voice, prompt messaging and video, amongst different functions.

SIP can be accountable for the implementation of different voice session associated capabilities corresponding to holding voice calls, transferring calls, or internet hosting a number of voice conversations concurrently (name conferencing).

SIP utilises the Session Description Protocol (SDP) to barter information varieties accessible at both finish. Voice and video information carrying is often carried out with the RTP and RTCP protocols.

MGCP (Media Gateway Management Protocol) – MGCP is a protocol sometimes used internally to methods to symbolize the entire system as a single entity. MGCP methods are made up of Name Brokers and Gateways. The Name Brokers preserve stateful data and offload the vast majority of the management work from the Gateway(s). Voice and video carrying is carried out by RTP and RTCP.

MEGACO/H.248 – MEGACO is similar to MGCP and is the usual used for Name Agent architectures.

IAX (Inter Asterisk eXchange) – The IAX protocol was developed by a crew of open-source builders engaged on the Asterisk undertaking, a very fashionable and profitable open-source PBX described later on this doc.

There may be not a lot trade help for IAX, nevertheless help has been slowly growing. There was an absence of excellent documentation for the IAX protocol, and lots of industrial distributors are hesitant to provide their help to a protocol not ratified by a requirements physique such because the IETF or ITU.

LTP (Light-weight Telephony Protocol) – the LTP is a binary light-weight protocol that’s NAT pleasant and based mostly completely on free codecs. It’s simply understood and in use since 1999.

VoIP Cellphone {Hardware}/Software program[edit]

VoIP {hardware} and software program implementations enable telephone calls to be positioned over IP. Software program which makes use of a pc’s soundcard gadget to supply voice enter and output is called a Softphone.

A number of {hardware} units exist to supply VoIP telephony. ATA units present customary telephone ports, permitting customary telephone handsets and different units corresponding to fax machines to make the most of VoIP companies. Specialised VoIP handsets are additionally accessible, usually offering superior options supported by VoIP gateways.

{Hardware} telephones typically present a greater high quality VoIP expertise than Softphones; nevertheless the comfort of not requiring further {hardware}, and the smaller expense concerned in Softphone merchandise make softphones a well-liked alternative.

Softphones[edit]

A softphone is a soundcard gadget to supply voice enter (by way of microphone) and sound output (by way of headphones/audio system) with out the necessity for further {hardware}. Handsets and headsets can be found which plug into the soundcard and resemble regular telephone headsets/handsets, making softphone use extra snug.

A preferred open supply softphone that runs on each Home windows and Linux platforms is ekiga which helps each H.323 and SIP protocols. Xten X-Lite is one other in style free Softphone which gives all essential options to provoke and obtain telephone calls by way of SIP. An inventory of among the softphones accessible will be discovered on voip-info.org right here, and on Wikipedia right here.

VoIP Handsets[edit]

ATA Adaptors[edit]

`ATA (Analog Phone Adaptor) Gadgets present the power to attach customary phone handsets to a {hardware} unit which gives VoIP functionality.

A bonus of ATA units is the power to attach different telephony units, corresponding to TTY units, FAX machines, Pay TV models and even modems.

VoIP Gateway Techniques[edit]

This chapter offers with the advanced world of VoIP gateway methods. These methods are accountable for offering some quantity of name management and routing from VoIP telephone units.

Every of those methods is accountable for receiving calls from a VoIP handset, ATA or softphone gadget by way of a supported VoIP protocol, consulting a dial-plan or different name routing desk, and appropriately routing the decision. Different options made accessible by some or all of those gateways embody:

Authentication – Making certain that shoppers have permission to make use of VoIP assets.

Name Accounting – Offering monitoring of calls by way of the gateway, together with reviews and value controls.

Advanced Dialplans – Most dialplans configured on easy units corresponding to ATAs and softphones solely enable for a really easy dialplan configuration. Utilizing a VoIP gateway corresponding to Asterisk, components corresponding to time of day, and even values obtained from exterior sources can be utilized to find out which path a name ought to take.

{Hardware} Integration – Most gateways present the power to interface with bodily {hardware} corresponding to FXS (inner line) and FXO (PSTN line) playing cards to create a PBX system. They moreover present the dialplan configuration to find out which calls ought to route by way of IP, and which ought to make the most of a card. It’s important when choosing gateway software program to guage which units they help, and guarantee yours is listed. Some units (for instance, Cisco line playing cards) might require the seller’s particular implementation of VoIP gateway for any help in any respect.

Inner Numbering – Some gateways provide the power to supply inner extension numbering for shoppers, permitting one VoIP gadget to dial one other gadget registered in your native gateway utilizing a brief inner extension quantity.

When would you employ a VoIP gateway?[edit]

Through which conditions would you employ a VoIP gateway product?

Usually, when you require a sophisticated configuration corresponding to superior name accounting for billing, value controls, advanced dialplans, or the power to name between extensions, it’s best to implement a gateway resolution.

If you’re excited about integrating further {hardware} corresponding to FXO playing cards (interfaces to the PSTN) you MUST use a gateway product.

If you’re designing VoIP options for mid-large measurement enterprise, or will likely be requiring IVR (Interactive Voice Response) performance, it’s best to implement a VoIP gateway.

Asterisk: The open-source PBX[edit]

Asterisk is a particularly in style open-source PBX system which runs on BSD, Linux, Mac OS X, and Home windows. The undertaking is sponsored by Digium, a PBX {hardware} producer.

Asterisk has help for ENUM, e911, Caller ID, all name controls corresponding to Forwarding, Conferencing, Maintain, Switch and Name Ready. Extra options corresponding to Name Monitoring, Name Recording and Privateness Controls additionally exist.

Along with the above, Asterisk is ready to present IVR capabilities, permitting interactive voice prompts, name queuing, and lots of many superior name routing options.

In all, Asterisk is a remarkably full featured commercial-grade PBX system accessible freed from cost.

Bayonne[edit]

OpenSER[edit]

Internet web page: http://openser.org

OpenSER is a sturdy and highly effective SIP server. Launched underneath GPL, OpenSER is the primary free server with built-in TLS, providing safe VoIP communications. It has an structure designed for scalability and suppleness and excessive performances.

Major traits:

  • SIP proxy/registrar/redirect server (RFC3261)
  • transaction stateful
  • UDP/TCP/TLS help
  • modular structure
  • scripting configuration file with pseudo-variables
  • authentication, authorization and accounting by way of database, radius or textual content recordsdata
  • enum help
  • NAT traversal system
  • formatted logging
  • least value routing
  • Name Processing Language (CPL)
  • MySQL/Postgres/Flat recordsdata database backend
  • sever monitoring

SIP Specific Router[edit]

Internet web page: http://www.iptel.org/ser

SIP Specific Router (SERi) is a high-performance, configurable, free SIP server licensed underneath the open-source GNU license . It could act as SIP (RFC 3261) registrar, proxy or redirect server. SER will be configured to serve specialised functions corresponding to load balancing or SIP front-end to software servers, SEMS for instance.

SER options:

  • full help of RFC 3261 performance,
  • a wide range of database backends (mysql, oracle, postgres, radius, text-db),
  • administration options (distant administration by way of XML-RPC, load-balancing),
  • NATi traversal, telephony options (LCR, speeddial),
  • multidomain internet hosting, ENUM, presence, and much more.

SER is moreover enhanced by a wide range of further SIP instruments, which offer performance for administration, media processing, CDRi processing, and so on.

SER is as we speak default a part of quite a few working methods and their distributions: Debian, FreeBSD, Gentoo, NetBSD, OpenBSD, OpenSUSE, Solaris.

SER historical past spans again to the earlier century. SER has been used since 2002 for varied totally different functions, continuously within the trade by main ISPs/ASPs and by universities to allow VoIPi companies. SER’s explicit energy is its efficiency (SER runs properly even underneath heavy load attributable to massive subscriber populations or irregular operational circumstances), flexibility (SER’s real configuration language and module interface enable excessive diploma of customization) and interoperability (examined and operated towards tens of SIP merchandise over time, together with however not restricted to (Microsoft, Cisco, Mitel, snom, Pingtel, Siemens, xten, and lots of others).

Yate – But One other Telephony Engine[edit]

Yate is a next-generation telephony engine; whereas at the moment targeted on Voice over Web Protocol (VoIP), its energy lies in its capacity to be simply prolonged. Voice, video, information and prompt messaging can all be unified underneath Yate’s versatile routing engine, maximizing communications effectivity and minimizing infrastructure prices for companies.

Yate can be utilized as a:

  • VoIP server ****
  • VoIP consumer
  • VoIP to PSTN gateway
  • PC2Phone and Phone2PC gateway
  • H.323 gatekeeper
  • H.323 a number of endpoint server
  • SIP session border controller
  • SIP router
  • SIP registration server
  • IAX server and consumer
  • IP Telephony server and consumer
  • Name heart server
  • IVR engine
  • Pay as you go and post-paid playing cards system

YXA[edit]

YXA is a SIP server written within the programming language Erlang [1] at Kungliga Tekniska Högskolan and Stockholms universitet. Erlang was developed by Ericsson to program abnormal phone switches, with the purpose of creating a programming system fault-tolerant and sturdy.

This helps YXA to be a sturdy SIP server/stack able to serving tens of hundreds of customers.
The tasks purpose is to make YXA complient to all RFC requirements related to SIP.

  • It’s RFC3261 compliant SIP-server, able to all the things a generic area wants :
    • Registrar that retains observe of your customers
    • Handles incoming SIP requests to your area
    • Handles routing of requests out of your customers to distant domains
    • TCP, UDP and TLS (together with SIPS) help
    • Routinely maps e-mail addresses of your customers to their SIP addresses, in case you have the e-mail addresses in LDAP
    • Handles a number of domains utilizing a single server occasion
  • ENUM help for PSTN-bypass each time potential
  • IPv6 help
  • Forking, each parallel and sequential
  • CPL (RFC3880) help for superior user-control of occasions (at the moment incoming calls solely)
  • Modular person database, at the moment with LDAP, Mnesia, MySQL and text-file backends
  • PSTN vacation spot entry management (per person or for nameless customers)

VoIP PABX Integration[edit]

Superior VoIP Configuration[edit]

Visitors shaping and bandwidth management[edit]

Accurately terminating emergency calls[edit]

Explicit consideration ought to be given to making sure that emergency calls on VoIP traces are dealt with appropriately. In some circumstances, it will not be potential to supply helpful or dependable emergency service by way of VoIP companies. On this case, it could be potential to dump emergency calls to a landline service.

If dependable emergency service isn’t accessible, the handset or VoIP gadget ought to clearly be marked with a warning.

Some VoIP suppliers enable emergency calls to be routed by way of their service; nevertheless that is solely of use if you’re inside an space lined by the receiving emergency name centre. Moreover, an absence of bandwidth or Web service or an influence loss can result in lack of emergency service by way of phone.

Ideas for coping with emergency calls[edit]

If a functioning landline handset is close by, it’s advisable to connect a discover to the VoIP gadget directing emergency calls to the landline. It’s a lot simpler for emergency companies to help if they’ve data in your location which is usually not accessible by way of VoIP calls.

The place calls are dealt with by a gateway administered by your self or your organisation, and the power to route sure calls by way of landline exists, the dialplan to your VoIP gateway ought to instantly route emergency calls by way of the landline, and the place potential, any energetic conversations ought to be dropped if there should not adequate traces to deal with the decision.

It is very important discover out whether or not your VoIP supplier(s) provide emergency numbers, what areas they can cowl and whether or not they can present enhanced emergency companies corresponding to e911 initiatives which permit suppliers to transmit data on subscribers to 911 name centres.

Be certain that emergency calls are routed by way of probably the most applicable gateways in your dial plan.

Incoming telecommunications companies[edit]

IVR (Interactive Voice Response) software[edit]

IVR functions are automated response models which obtain enter from the person (historically within the type of DTMF) and permit the person to progress alongside a pre-set interactive path. For instance, a menuing system the place the IVR might announce Press 1 for Gross sales, 2 for Marketting after which await person enter. As soon as the person enter arrives the IVR unit might patch by means of the decision to a different line and/or might present a pre-recorded message.

ENUM/E164 Expertise[edit]

The next part is kind of technical, because it offers with sophisticated infrastructure and the mixing of VoIP into core web applied sciences.

The ENUM protocol (described in RFC doc RFC 3761) interprets customary phone quantity routing (often known as E.164, because the ITU defines the usual by which phone numbers are assigned within the E.164 customary) into an addressing scheme in a position to path to VoIP (eg. SIP/H323) gateways utilizing DNS expertise.

ENUM expertise permits VoIP servers to make use of the DNS infrastructure to question routing paths for terminating voice calls given a telephone quantity.

When a name is initiated to a given telephone quantity, a VoIP gateway able to ENUM service will question the supplied quantity in DNS to find out whether or not there’s a path apart from the PSTN line accessible to the vacation spot service.

This permits suppliers to promote paths by way of VoIP to companies in any other case solely accessible by way of a VoIP to PSTN gateway.

ENUM information might include different data along with a VoIP gateway handle, corresponding to an internet URL or e-mail handle.

Personal vs Public ENUM databases[edit]

The general public ENUM database is a set of ENUM information, often known as NAPTR RRs in DNS converse, which map phone numbers to VoIP service gateways. These information are saved in a hierarchical tree beneath the e164.arpa area.

Data inside the e164.arpa area are delegated to their accountable proprietor by way of the E.164 addressing conference. Very similar to the in-addr.arpa area, which is used to supply mapping of IP addresses to names, these information are delegated in blocks to the company accountable for this quantity project.

The company is then accountable for additional smaller delegations the place essential. Consequently, it’s unlikely that people or small companies may have entry to supply ENUM information for his or her phone numbers.

It additionally stands to motive that phone corporations can be hesitant to permit phone calls destined to their prospects to as a substitute terminate over the Web, as a lot of a telephone firm’s income can be derived from name termination charges, charged to carriers for calls terminating on an organization’s community.

That is unlucky for finish customers. By way of the ENUM database, you’d have the chance to direct VoIP-based calls to terminate over the Web by way of your VoIP gateway, slightly than by way of a telephone line.

Because of this, a number of free ENUM databases have been created to permit customers to share alternate IP-based paths to their telephone numbers. Most ENUM conscious gateways will enable further ‘search domains’ to be configured, offering entry to those unofficial databases.

While configuring a gateway to question these further domains can dramatically cut back prices by terminating calls over IP, subsequently liberating copper traces/voice channels for each the caller and recipient, there is a vital issue to think about. If a malicious or unauthenticated entry is added, an attacker can direct requires a telephone quantity to an arbitrary VoIP gateway.

Contemplate the instance of rival companies. If Enterprise A had been tech-savvy and entered their competitor’s quantity right into a public listing, directing calls to their very own VoIP gateway, any calls positioned by way of a VoIP gateway configured to question that listing can be directed to Enterprise A’s telephone, slightly than the supposed recipient, Enterprise B.

A few of these directories require voice verification of possession if you register telephone numbers. Others don’t require verification or have dispute decision insurance policies which take care of disputes on a case-by-case foundation.

One other much less critical however nonetheless vital consideration is the likelihood that your data will likely be harvested from the database, or offered by the database proprietor. ENUM data could also be useful to advertisers or telephone spammers, as it might present them with a listing of legitimate telephone numbers, and permit them to name totally free by way of IP.

Public ENUM registries[edit]

VoIP Suppliers Worldwide[edit]

VoIP expertise has utterly modified how corporations work together with their prospects. VoIP has scaled a lot that now it’s getting used for residential in addition to enterprise functions. VoIP suppliers are primarily of two varieties, Retail VoIP supplier and wholesale VoIP supplier[1]. There are bunch of VoIP suppliers on-line however these are among the greatest VoIP suppliers on-line.

ENUM Trial Registries[edit]

Many ENUM trial registries have gotten accessible. Most of those registries are regional, and infrequently are restricted by telephony supplier, restricted variety of trial members, or a trial charge.

Australian ENUM Trial – Charge applies for registration

Austrian ENUM Trial – Open to Austrian residents, a charge applies for registration.

German ENUM Trial

Japanese ENUM Trial – Open to ETJP contributors since 2003. Presently has 46 members.

Taiwan ENUM Trial

UK ENUM Trial

Appendices[edit]

Appendix A: Glossary of Phrases[edit]

ATA Analog Phone Adaptor
Broadband Utilizing a number of transmissions concurrently over a single medium.
Baseband Utilizing a single transmission over a single medium. CAS Channel Related Signaling
CCS Widespread Channel Signaling
DTMF Twin-Tone Multi-Frequency
DS0 64Ok Channel
DSL Digital Subscriber Line
E1 32 DS0 Channels
FXO International eXchange Workplace
FXS International eXchange Subscriber
IP Web Protocol – IP is the frequent protocol, or language, utilized by all units on the Web to speak no matter vendor.
MGCP Media Gateway Management Protocol
NAT Community Handle Translation – The interpretation of a Community (IP) Handle to a different handle by a gateway or router gadget
PAT Port Handle Translation – The interpretation of a number of inside IP addresses to a single IP handle on the skin, utilized in typical residence functions.
PBX Personal Department Trade
PoE Energy over Ethernet
POTS Plain Outdated Phone System. See PSTN beneath.
PSTN Public Switched Phone Community
Q.931 Layer three ISDN Signaling
Q.921 Layer 2 ISDN Signaling
RFC Request For Feedback
RTCP Actual-Time Management Protocol
RTP Actual-Time Transport Protocol
SIP Session Initiation Protocol
T1 24 DS0 Channels
  1. Wholesale VoIP Supplier

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