You need Broadband internet connection. The more call volume you have the higher the bandwidth you will need. We have low call volume and are using a 7Mbps/700Kbps connection, this has been plenty of bandwidth for us. Wired Ethernet router. You have a ton of options here but your budget will likely decide this one.
Security for VoIP uses industry standard encryption technology such as SSL and VPN.
To connect VoIP phones to traditional telephony networks, you need to install an Analaog Telephone Adapter (ATA) which converts the analog signal into digital data.
A dial-up connection can support VoIP, but it is recommended to use broadband since certain codecs require higher bandwidths for quality purposes.
VoIP is fully compatible with calls to/from PSTN lines.
Depending on bandwidth quality and availability, VoIP quality is excellent.
A VoIP gateway is the means of converting telephony traffic into IP for transmission over a network.
Sound quality on LAN is excellent and a standard feature of PBXware.
In comparison to Public Switched Telephone Network (PSTN) services, switching to VoIP with significantly reduce your business telephone costs.
In the event of power failure, PBXware will continue to operate should it have an Uninterruptible Power Supply device installed in the system. This has the ability to maintain operations for minutes/hours until power is restored.
The benefits of PBXware are associated with cost, simplicity, efficiency and reliability. From saving money on telecommunications to keeping employees connected remotely, quality control to scalability, PBXware represents a new level of telephony efficiency.
PBXWare offers a number of benefits to your business, aiding cost savings, productivity, and efficiency. Among those features available: – Conference calling – Call recording – Call forwarding – Call waiting – Direct Inward Dialling – Interactive Voice Response – Music On Hold – Destinations Permissions – Backup – Automatic updates
PBXware enables business locations to communicate through VoIP, reducing costs. In addition, remote workers can also access these same networks, eliminating travel costs and time, while the ability to divert to mobile phones enhances flexibility. There is also the benefit of working with a browser-based administration system, reducing expenses on system maintenance, technical support, training, etc.
PBXware operates on a Linux OS platform.
Emergency calls can be placed by direct dialling, or with a prefix number for an outgoing phone followed by the emergency number. – Select your primary, secondary, and tertiary trunks for each destination – On placing a call to any of the configured destinations, PBXware will attempt to connect using first the primary trunk, then the secondary, and finally the third, depending on performance issues
The PBXware system administrator offers easy navigation through configuration, with only a few clicks of a mouse to activate the business features you need.
Accessing: – web GUI – TCP 80, 443, 81 – ssh – TCP 2020 For SIP phones: – TCP 10001, 5060-5069 – UDP 5060-5069, 10000-20000 For IAX phones: – TCP 5038, 5037 – UDP 4569 For Jabber: – TCP 5222 or 5223
Session Initiation Protocol (SIP) is a telephony signalling protocol used to establish, modify and terminate VoIP telephone calls.
Session Description Protocol (SDP) is a format for describing streaming media content initialisation parameters.
Echo cancellation removes echo from a voice communication in order to improve voice call quality and reduce bandwidth consumption. Echo cancellation is required since speech compression techniques and packet processing delays generate two types of echo: acoustic and hybrid.
Real Time Transport Protocol (RTP) defines a standard packet format for delivering audio and video data over the internet.
RTP Control Protocol (RTCP) works with RTP to send control packets to call participants. The primary function is to provide feedback on the quality of service provided by RTP.
Put simply, a SIP URI is a user’s SIP phone number, but resembles an email address in appearance. For example, the structure is: sip:x@y:Port (where x=username and y=host)
SIP Methods and Requests are the means with which a call session is established.
INVITE = Establishes session ACK = Confirms an INVITE BYE = Ends session CANCEL = Cancels establishing a session REGISTER = Communicates user location (host name, IP) OPTIONS = Communicates information about the capabilities of the calling and receiving SIP phones
-1xx = Informational responses, such as 180, which means ringing -2xx = Success responses -3xx = Redirection responses -4xx = Request failures -5xx = Server errors -6xx = Global failures 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. Proxies should use proxy authorisation -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 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
A SIP call session between two phones is established as follows: -The calling phone sends out an invite. -The called phone sends an information response: 100 – Trying. -When the called phone starts ringing a response is sent: 180 – Ringing. -When the caller picks up the phone, the called phone sends a response: 200 – OK. -The calling phone responds with: ACK – acknowledgement. -Now the actual conversation is transmitted as data via RTP. -When the person calling hangs up, a BYE request is sent to the calling phone. -The calling phone responds with: 200 – OK.
To deal with fax, set PBXware options thus: -Connect phone/fax line to PBXware box -Create a trunk for this line -Create a new DID and point it to the fax destination Once the fax enters the DID, PBXware will accept its signal and receive the fax. The same will be converted to a PDF and emailed to administrator.
Codecs convert analog signals to digital. This is needed for voice transmission over a network. The following codecs are supported by PBXware: – GSM – 13Kbps (full rate) – iLBC – 15Kbps size – ITU G.711 – 64Kbps (ulaw|alaw) – ITU G.722 – 48/56/64Kbps – ITU G.723.1 – 5.3/6.3Kbps – ITU G.726 – 16/24/32/40Kbps – ITU G.728 – 16Kbps – ITU G.729 – 8Kbps – Speex – 2.15 to 44.2Kbps – LPC10 – 2.5Kbps – DoD CELP – 4.8Kbps
FoIP stands for Fax over Internet Protocol and refers to the process of transmitting faxes over a VoIP network. FoIP works via T38 (compatible with PBXware) and requires a T38 capable VoIP gateway as well as a T38 capable fax machine, fax card or fax software.
PBXware consists of one or more SIP/VoIP phones and an optional VoIP Gateway. Those with soft or hardware based phones register with the PBXware server to establish connections to make calls. A PBXware system features a directory of all users and is able to connect an internal call or route an external call via VoIP gateway or a VoIP service provider.
VoIP phones are available in a number of forms including VoIP softphones, USB phones, hardware SIP phones, and analog phones with an ATA adapter.
FXS and FXO are the name of ports used by analog phone lines. Foreign eXchange Subscriber (FXS) is a port that delivers the analog line to the subscriber; and Foreign eXchange Office (FXO) is a port that receives the analog line. FXO and FXS are always paired, similar to a male/female plug.
Simple Traversal of User Datagram Protocol Through Network Address Translators (STUN). The STUN server enables a client to find out its public address, the NAT they are behind, and the internet side port associated by the NAT with a particular local port. This information is used to establish communications between the client and the VoIP provider.
The main component of an IP PBX, the SIP server handles the setup of all calls in the network. PBXware is an example of a SIP server.
ENUM stands for Telephone Number Mapping, linking a phone number to an internet address published in the DNS system. The owner of an ENUM number can publish where a call should be routed to via DNS entry. Different routes can be defined for different types of calls.