The public switched telephone network (PSTN) is the aggregate of the world's circuit-switched telephone networks that are operated by national, regional, or local telephony operators, providing infrastructure and services for public telecommunication. The PSTN consists of telephone lines, fiber optic cables, microwave transmission links, cellular networks, communications satellites, and undersea telephone cables, all interconnected by switching centers, thus allowing any telephone in the world to communicate with any other. Originally a network of fixed-line analog telephone systems, the PSTN is almost entirely digital in its core and includes mobile as well as fixed telephones


One Way Audio (OWA) is a common VoIP issue, occurring when one device in a call can not receive audio but can still send, or vise-versa. The causes of OWA can vary from firmware or software issues with the phone, to firewall or network issues with the data streams


Direct inward dialing (DID) service has similar relevance for Voice over Internet Protocol (VoIP) communications. To reach users with VoIP phones, DID numbers are assigned to a communications gateway connected by a trunk to the public switched telephone network (PSTN) and the VoIP network. The gateway routes and translates calls between the two networks for the VoIP user. Calls originating in the VoIP network will appear to users on the PSTN as originating from one of the assigned DID numbers


The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks. In VoIP, we use this protocol to begin and end calls


The Real-time Transport Protocol (RTP) defines a standardized packet format for delivering audio and video over IP networks. RTP is used extensively in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications, television services and web-based push-to-talk features


A wide area network (WAN) is a network that covers a broad area (i.e., any telecommunications network that links across metropolitan, regional, or national boundaries) using private or public network transports. This is the connection to the outside world


A local area network (LAN) is a network that interconnects computers/phones in a limited area such as a home, school, computer laboratory, or office building using network media


A private branch exchange (PBX) is a telephone exchange that serves a particular business or office, as opposed to one that a common carrier or telephone company operates for many businesses or for the general public


Network Address Translation (NAT) is the process of modifying IP address information in IPv4 headers while in transit across a traffic routing device


The User Datagram Protocol (UDP) uses a simple transmission model with a minimum of protocol mechanism. It has no handshaking dialogues, and thus exposes any unreliability of the underlying network protocol to the user's phone. As this is normally IP over unreliable media, there is no guarantee of delivery, ordering or duplicate protection. UDP provides checksums for data integrity, and port numbers for addressing different functions at the source and destination of the datagram.

The audio portion of your call will be sent in UDP packets due to the real-time needs of audio


The Transmission Control Protocol (TCP) provides reliable, ordered, error-checked delivery of a stream of octets between programs running on computers or phones connected to a local area network, intranet or the public Internet. It resides at the transport layer


The Domain Name System (DNS) is a hierarchical distributed naming system for computers, services, or any resource connected to the Internet or a private network. It translates easily memorized domain names to the numerical IP addresses needed for the purpose of locating computer services and devices worldwide


Indicates a client is being invited to participate in a call session


Confirms that the client has received a final response to an INVITE request


Terminates a call and can be sent by either the caller or the callee


Cancels any pending request.


Registers the address listed in the To header field with a SIP server


Provisional acknowledgment


The Primary Rate Interface (PRI) is a standardized telecommunications service level within the Integrated Services Digital Network (ISDN) specification for carrying multiple DS0 voice and data transmissions between a network and a user


The T1 (or T-1) carrier is the most commonly used digital transmission service in the United States, Canada, and Japan. In these countries, it consists of 24 separate channels using pulse code modulation (PCM) signals with time-division multiplexing (TDM) at an overall rate of 1.544 million bits per second (Mbps)


An Integrated Access Device (or IAD) is a customer premise device that provides access to wide area networks and the Internet. Specifically, it aggregates multiple channels of information including voice and data across a single shared access link to a carrier or service provider PoP (Point of Presence). The access link may be a T1 line, a DSL connection, a cable (CATV) network, a broadband wireless link, or a metro-Ethernet connection


A Session Border Controller (SBC) is a device regularly deployed in Voice over Internet Protocol (VoIP) networks to exert control over the signaling and usually also the media streams involved in setting up, conducting, and tearing down telephone calls or other interactive media communications

SIP Trunk

SIP trunking is a Voice over Internet Protocol (VoIP) and streaming media service based on the Session Initiation Protocol (SIP) by which Internet telephony service providers deliver telephone services and unified communications to customers equipped with SIP-based private branch exchange (IP-PBX) and Unified Communications facilities

Packet Loss

When one or more packets of data traveling across a network fail to reach their destination


A synonym for delay, Latency is an expression of how much time it takes for a packet of data to get from one designated point to another


The variation in the time between packets arriving, caused by network congestion, timing drift, or route changes

100 Trying

Extended search being performed may take a significant time so a 100 Trying response is sent

180 Ringing

Destination user agent received INVITE, and is alerting user of call (the phone is now ringing)

183 Session in Progress

This response may be used to send extra information for a call which is still being set up.

200 OK

Indicates the request was successful. If this was in response to an call INVITE, the call is now established.

302 Moved Temporarily

You will see this whenever a call comes into our network and the number resides on the other server. The call will be moved over to the other server/SBC for completion.

401 Unauthorized

This will be the code kicked back by a server to any request that wasn't cleared or authenticated. If you see this error when registering devices, you should ensure the SIP authentication settings in the phone match what you have currently provisioned in Broadsoft.

403 Forbidden

The server understood the request, but is refusing to fulfill it. This can happen when the phone is trying to authenticate with a SIP Authorization Name/Pass that doesn't match what is currently provisioned for that user in Broadsoft.

404 Not Found

The server has definitive information that the user does not exist at the domain specified in the Request-URI. You will see this if your phone is trying to register with an invalid SIP User Name

482 Merged Request

The server was expecting something else, and your phone sent the same packet as it did earlier instead. This is typically caused by a Firewall blocking packets in from our server,

486 Busy Here

The number you dialed is Busy.

487 Request Terminated

Request has terminated by bye or cancel

604 Does Not Exist Anywhere

The server has authoritative information that the requested user does not exist anywhere


Dual-tone multi-frequency signaling (DTMF) is used for telecommunication signaling over analog telephone lines in the voice-frequency band between telephone handsets and other communications devices and the switching center. The version of DTMF that is used in push-button telephones for tone dialing is known as Touch-Tone

Have more questions? Submit a request


Powered by Zendesk