Question 1

What kind of internet connection and bandwidth do I need for SIP trunking?

Accepted Answer

For SIP trunking you need a broadband internet connection sized to your busy-hour concurrent calls rather than to your total number of phones, with roughly 85 to 90 kilobits per second in each direction for every simultaneous call using the common G.711 codec, or about 31 kilobits per second per call with the more compressed G.729a.[1][2][6] Multiply your peak concurrent calls by the per-call figure and add 10 to 20 percent of headroom for signaling and other traffic to get the floor for the connection.[7][8] Raw speed is only half of it, because the connection also has to hold one-way latency under about 150 milliseconds, jitter under about 30 milliseconds, and packet loss under roughly 1 percent, and voice degrades on delay and loss well before it runs out of raw bandwidth.[5][9] Upload capacity is usually the limiting factor on asymmetric consumer lines, so a symmetric business connection with quality-of-service prioritization, often fiber, is the typical recommendation.[9][11]

Question 2

What equipment do I need to use SIP trunking?

Accepted Answer

To use SIP trunking you need a SIP-capable phone system, endpoints for the people placing calls, an internet connection sized for voice, and a session border controller at the network edge, plus a gateway only if you are keeping analog or legacy equipment.[1][9][14] The phone system is usually an IP-PBX, either an appliance or software you run on-site or a hosted PBX the provider operates for you, and it speaks the Session Initiation Protocol to set up and tear down each call.[1][2] The people on the calls use IP desk phones or softphones running on a computer or mobile device, and hardware phones commonly draw power over the same Ethernet cable that carries their traffic.[4] A session border controller sits between your system and the provider to secure the SIP session, reconcile mismatched settings, and act as a back-to-back user agent on both call legs.[7][8] If you still have analog phones, fax machines, or a PRI-era PBX, an analog telephone adapter or VoIP gateway converts that traffic to SIP so it can ride the trunk.[5][6]

Question 3

Can I use SIP trunking with my existing phone system?

Accepted Answer

Yes, in most cases you can keep your existing phone system and connect it to a SIP trunk, though what that takes depends on the equipment you already run.[2] A phone system that speaks the Session Initiation Protocol natively, which covers most IP-PBX platforms sold in the last decade, connects to a provider by entering the trunk credentials in the PBX, sometimes with a session border controller at the edge.[1][7] Older equipment still works: a digital PBX on PRI or T1 lines connects through a gateway that converts SIP and its RTP audio into the circuit signaling the PBX expects, and analog phones or a key system connect through an adapter that presents them with a normal dial tone.[4][5][6] Because the provider now carries your calls to the public phone network, the service falls under the FCC's interconnected VoIP rules for 911 and registered caller location, regardless of how old your phone system is.[9][10]

Question 4

How many simultaneous calls do I need or get on a SIP trunk?

Accepted Answer

The number you need is the count of calls your busiest hour runs at the same time, not the number of employees, phones, or numbers you hold, and each of those calls occupies one channel, or SIP session, on the trunk.[1][5] A common starting point for a general office is one channel for every three to four employees, while a contact center needs closer to one channel per active agent, but the accurate way to size it is to feed your busy-hour call volume and average call length into an Erlang B calculation at a one percent blocking target.[4][5][2][3] The number you actually get is not fixed by the trunk itself: capacity is limited by your internet bandwidth, your codec, and any per-plan cap your provider sets, since each concurrent call consumes roughly 80 to 115 kilobits per second once packet overhead is counted.[6][7][5] As a rough ceiling, a clean 2 megabit connection carries about 25 G.711 calls, and providers can raise or lower your channel count in software rather than installing new circuits.[6][4][11]

Question 5

What is a SIP trunk channel?

Accepted Answer

A SIP trunk channel is the capacity to carry one phone call at a time over a SIP trunk, which is why a channel is usually described as a single concurrent call path or the digital equivalent of one traditional phone line.[5][7] Each active call, inbound or outbound, occupies one channel for its duration and frees it the moment the call ends, so the number of channels you buy sets the ceiling on how many calls can happen at the same time.[6][7] Channels are counted separately from phone numbers, and a business can hold far more direct inward dial numbers than channels because most numbers sit idle at any given moment while only a fraction ring at once.[6] The term carries over from the digital PRI circuit, where a T1 provided 23 voice channels, but on a SIP trunk a channel is a software session limited mainly by your internet bandwidth rather than by physical wiring.[4][13] Some providers still sell fixed channel bundles, while elastic services place no hard cap on concurrent calls and instead throttle how many new calls you can start each second.[12]

Question 6

What is the difference between SIP trunking and cloud PBX?

Accepted Answer

SIP trunking and cloud PBX differ in what the provider actually gives you: a SIP trunk is a connectivity service that links a phone system you already own to the public telephone network, while a cloud PBX is a complete phone system that a provider hosts and runs for you over the internet.[2][3][5] With SIP trunking you keep and manage your own private branch exchange, and the trunk only carries calls in and out, so every calling feature comes from your system and your IT staff handle its upkeep.[2][6] A cloud PBX, also called a hosted PBX, moves the whole system into the provider's data centers, bundles routing, voicemail, and unified communications features into a per-user subscription, and shifts maintenance and updates to the provider.[3][4] The two are priced on different units, roughly 15 to 30 dollars per concurrent channel for a trunk against 15 to 50 dollars per user each month for a hosted system, and they often work together, since a cloud PBX commonly uses SIP trunks inside its own network.[4][8] Both fall under the FCC's interconnected VoIP rules for 911 once they reach the public phone network.[9][10]

Question 7

What is the difference between SIP trunking and PRI?

Accepted Answer

The difference between SIP trunking and PRI is the technology that carries your calls: a PRI is a physical, dedicated digital circuit that uses time-division multiplexing over a T1 line, while a SIP trunk is a virtual voice service that sends calls as data packets over your internet connection using the Session Initiation Protocol.[1][2] A North American PRI carries a fixed 23 voice channels plus one signaling channel on each circuit, so adding capacity means ordering another circuit and waiting weeks for installation, whereas a SIP trunk counts capacity as concurrent call paths a provider can raise or lower in software.[1][4][5] SIP trunking usually costs less because you pay for the channels you actually use instead of a full circuit year-round, and it does not depend on the copper and TDM infrastructure that carriers are now retiring.[4][7][10] The main things a PRI still does better are running through a local power outage and delivering fixed quality that does not compete with your data traffic, since a SIP trunk depends on the internet connection underneath it.[4]

Question 8

What is the difference between SIP trunking and VoIP?

Accepted Answer

VoIP and SIP trunking are not competing products, because VoIP is the broad category and SIP trunking is one specific service inside it. VoIP, short for voice over Internet Protocol, is the general set of technologies that carry voice as data across IP networks rather than over the traditional phone network.[2] SIP trunking is one way to deliver that: a service that uses the Session Initiation Protocol to connect a business's own phone system, an IP-PBX, to the public telephone network.[1][3] Every SIP trunk is VoIP, but not every VoIP service uses SIP, since VoIP can also run on other protocols such as H.323 or MGCP and on fully hosted systems that need no trunk at all.[2][4] The practical decision most buyers face is a SIP trunk, which keeps your existing phone system and links it outward, against a hosted VoIP service, which moves the whole phone system into the provider's cloud.[5][9]

Question 9

How does SIP trunking work?

Accepted Answer

SIP trunking works by splitting a phone call into two jobs that both travel over your internet connection: the Session Initiation Protocol sets up and later tears down the call, while the audio moves separately as Real-time Transport Protocol packets.[2][6] When your IP-PBX places a call, it uses DNS to locate the provider's servers, authenticates over the trunk by a trusted IP address or a username-and-password digest, then sends a SIP INVITE that offers the codecs and media ports it wants to use.[7][10][3] The provider answers with the codecs and ports it accepts, the two ends exchange ringing and answer messages, and once they agree the voice flows as RTP until a BYE ends the session.[3][4][6][2] Because the provider joins that trunk to the public switched telephone network, the call reaches any landline or mobile number, which also places the service under the FCC's interconnected VoIP rules for 911.[1][12][13]

Question 10

What is SIP trunking?

Accepted Answer

SIP trunking is a way to deliver phone service over an internet connection instead of over dedicated telephone lines, using the Session Initiation Protocol to set up and tear down calls between a business phone system and a provider that connects to the public telephone network.[1][2] The trunk itself is virtual: an internet telephony service provider carries your calls in software, so a company with an IP-PBX no longer needs the physical T1 or PRI circuits that older systems required.[1][6] SIP handles the call signaling while the voice travels separately as RTP media packets, and a session border controller usually sits at the network edge to secure and normalize the traffic.[2][3][5] Because capacity is measured in concurrent call paths rather than physical lines, you can add or remove channels without installing hardware, and a service that reaches the regular phone network falls under the FCC's interconnected VoIP rules for 911 and caller location.[6][10][11]

SIP Trunking