Glossary / SIP Protocol
🔗 SIP Protocol covers the signalling language that Cloud PBX systems use to set up, manage, and end phone calls over the internet. This section contains 38 terms, from SIP methods like INVITE and BYE to response codes like 200 OK and 503 Service Unavailable.
On this page: SIP · SIP URI · SIP Request/Method · INVITE · BYE · REGISTER · ACK · CANCEL · OPTIONS · REFER · SUBSCRIBE/NOTIFY · INFO · MESSAGE · UPDATE · PRACK · Re-INVITE · SIP Proxy · SIP Registrar · SIP Redirect Server · SDP · SIP Dialog · SIP Transaction · Call-ID · SIP Forking · Via Header · Contact Header · Record-Route Header · CSeq · Early Media · 100 Trying · 180 Ringing · 183 Session Progress · 200 OK · 302 Moved Temporarily · 401/407 Authentication Required · 403 Forbidden · 404 Not Found · 408 Request Timeout · 480 Temporarily Unavailable · 486 Busy Here · 487 Request Terminated · 488 Not Acceptable Here · 500 Server Internal Error · 503 Service Unavailable
SIP (Session Initiation Protocol)
A signalling protocol used to start, modify, and end communication sessions over the internet. SIP handles the setup of voice calls, video calls, and messaging between devices. It does not carry the actual voice data; it only manages the call connection. SIP is the foundation of nearly all modern Cloud PBX systems.
Related: SIP Trunking · VoIP · SDP
SIP URI
The address format used to identify a SIP user or device, similar to an email address. A typical SIP URI looks like sip:user@domain.com. It tells the network where to send a call or message. SIP URIs can also include a port number or transport type for more specific routing.
Related: DID / DDI · SIP Registrar · Contact Header
SIP Request / Method
A command sent from one SIP device to another to perform a specific action. Each request type has a name (like INVITE or BYE) and tells the receiving side what the sender wants to do. The most common methods handle starting calls, ending calls, and registering devices. SIP defines a core set of methods, with extensions adding more over time.
Related: INVITE · BYE · REGISTER · SIP Transaction
INVITE
The SIP method used to start a new call or session. When you dial a number, your phone sends an INVITE to the other party. The INVITE contains details about what type of media (voice, video) the caller supports. It is the most important SIP method and the starting point of every call.
Related: SDP · ACK · Re-INVITE · SIP Dialog
BYE
The SIP method used to end an active call. Either party in a call can send a BYE to hang up. Once the other side confirms with a 200 OK response, the call is fully terminated. BYE can only be sent after a call has been established (after the initial INVITE is answered).
Related: INVITE · 200 OK · CANCEL
REGISTER
The SIP method a phone or device uses to tell the system where it can be reached. When your desk phone starts up, it sends a REGISTER message to the registrar server with its current IP address. This registration expires after a set time and must be renewed periodically. Without registration, incoming calls cannot find your device.
Related: SIP Registrar · Trunk Registration · Contact Header
ACK
A SIP method sent to confirm that an INVITE has been successfully completed. After the called party answers (200 OK), the caller sends an ACK to acknowledge receipt. This three-way handshake (INVITE, 200 OK, ACK) ensures both sides agree the call is established. ACK is the only SIP method that does not receive its own response.
Related: INVITE · 200 OK · SIP Transaction
CANCEL
A SIP method used to stop an INVITE that has not yet been answered. If you hang up while the phone is still ringing, your device sends a CANCEL. It tells the network to stop trying to connect the call. CANCEL only works before the other party picks up; once answered, you must use BYE instead.
Related: INVITE · BYE · 487 Request Terminated
REFER
A SIP method used to transfer a call to another party. When you initiate a call transfer, your phone sends a REFER to the other party, instructing them to contact a new destination. The transfer can be attended (you speak to the new party first) or blind (the call moves immediately). REFER is the mechanism behind most call transfer features.
Related: Call Transfer · SUBSCRIBE/NOTIFY
SUBSCRIBE / NOTIFY
Two related SIP methods for event notifications. SUBSCRIBE tells a server "notify me when something changes" (for example, when a colleague's phone status changes). NOTIFY delivers those updates. Together, they power features like presence indicators (showing who is available) and voicemail waiting lights.
Related: Presence · REFER
INFO
A SIP method used to send information during an active call without changing the call itself. Common uses include sending DTMF tones (keypad presses) mid-call, for example when navigating an automated phone menu. INFO carries data within an existing dialog and does not affect the media stream.
Related: IVR · SIP Dialog
MESSAGE
A SIP method used to send instant text messages between SIP devices. Unlike INFO, MESSAGE can be sent outside of an active call. It works like SMS but travels over the SIP network. Some business phone systems use MESSAGE for internal team chat or short notifications.
Related: SIP URI · SIP Request/Method
UPDATE
A SIP method used to change session parameters (like media settings) before a call is fully established. While Re-INVITE changes settings during an active call, UPDATE does so during the setup phase. For example, it can modify the codec or add video while the phone is still ringing.
Related: Re-INVITE · SDP · Early Media
PRACK (Provisional Response Acknowledgment)
A SIP method used to reliably acknowledge provisional responses (like 180 Ringing). Standard SIP does not guarantee that provisional responses are received. PRACK adds reliability by requiring confirmation of each provisional response, which is important for features that depend on Early Media or ringback tones over unreliable networks.
Related: 180 Ringing · 183 Session Progress · Early Media
Re-INVITE
An INVITE sent within an already established call to change something about the session. Common uses include putting a call on hold (by changing the media direction), adding video to a voice call, or switching codecs. The other party can accept or reject the change. Re-INVITE does not create a new call; it modifies the existing one.
Related: INVITE · UPDATE · SDP · Call Hold
SIP Proxy
A server that routes SIP messages between devices. When you make a call, the INVITE goes to a proxy server first. The proxy looks up where the other party is registered and forwards the message. Proxies can also apply rules, such as routing calls based on time of day or blocking certain numbers. Most Cloud PBX systems use multiple proxy servers for reliability.
Related: SIP Registrar · SIP Redirect Server · Outbound Proxy
SIP Registrar
A server that keeps a directory of which devices are currently online and where they can be reached. When a phone sends a REGISTER message, the registrar stores the mapping between the user's SIP URI and the device's current IP address. When someone calls that user, the system checks the registrar to find the right device.
Related: REGISTER · SIP URI · SIP Proxy
SIP Redirect Server
A server that responds to a SIP request by telling the sender to try a different address. Instead of forwarding the call itself (like a proxy), it sends back a 3xx response (such as 302 Moved Temporarily) with the new address. The sender then contacts the new address directly. Redirect servers reduce the load on the network by removing themselves from the call path.
Related: SIP Proxy · 302 Moved Temporarily · SIP Forking
SDP (Session Description Protocol)
A format used to describe the media details of a call. When two devices set up a call, they exchange SDP information inside SIP messages. SDP specifies which codecs to use, the IP addresses for media streams, and whether the session includes voice, video, or both. SDP is not a standalone protocol; it travels inside SIP INVITE and response messages.
SIP Dialog
A long-running relationship between two SIP endpoints, established by an INVITE and identified by a unique combination of Call-ID, From tag, and To tag. All messages within the same call (re-INVITEs, BYE) share the same dialog. The dialog tracks the state of the call from setup to termination.
Related: Call-ID · INVITE · BYE · SIP Transaction
SIP Transaction
A single request-response exchange within SIP. For example, an INVITE followed by a 200 OK is one transaction. A BYE followed by a 200 OK is another. A dialog can contain multiple transactions. Transactions ensure that each request is matched with its correct response, even when many messages are in flight.
Related: SIP Dialog · CSeq · SIP Request/Method
Call-ID
A unique identifier assigned to every SIP call. The Call-ID is a text string included in every SIP message related to that call. It allows all devices and servers in the network to associate messages with the correct call session. Call-IDs are generated by the device that initiates the call and remain the same for the entire dialog.
Related: SIP Dialog · Via Header
SIP Forking
When a SIP proxy sends an INVITE to multiple destinations at the same time. For example, if a user has a desk phone and a mobile app registered under the same number, the proxy forks the INVITE to both devices. Whichever device answers first takes the call, and the other receives a CANCEL. Forking is how "ring all devices" features work.
Related: SIP Proxy · CANCEL · Ring Group
Via Header
A SIP header that records the path a message has taken through the network. Each proxy server that handles the message adds its own Via entry. Responses follow the Via path in reverse to get back to the sender. Via headers help with troubleshooting by showing exactly which servers processed a call.
Related: Record-Route Header · SIP Proxy
Contact Header
A SIP header that provides the direct address where a device can be reached. When a phone registers, the Contact header tells the registrar the device's current IP and port. During a call, the Contact header allows the other party to send subsequent messages directly, bypassing proxy servers if possible.
Related: REGISTER · SIP Registrar · SIP URI
Record-Route Header
A SIP header added by proxy servers that want to stay in the path of all future messages for a dialog. Without Record-Route, subsequent messages in a call might bypass the proxy and go directly between endpoints. Proxies that need to monitor, bill, or apply policy to calls use Record-Route to ensure they see every message.
Related: Via Header · SIP Proxy · SIP Dialog
CSeq (Command Sequence)
A SIP header that contains a sequence number and the method name for each request. It helps both sides keep track of the order of requests and match responses to the correct request. Each new request within a dialog increments the CSeq number, preventing confusion when multiple requests overlap.
Related: SIP Transaction · SIP Dialog
Early Media
Audio or video that plays before a call is fully answered. Common examples include custom ringback tones, pre-recorded announcements ("Please hold, your call is being connected"), or IVR menus that start before the called party picks up. Early Media is carried using 183 Session Progress responses with SDP information.
Related: 183 Session Progress · PRACK · SDP
180 Ringing
A SIP provisional response indicating that the destination phone is ringing. When you hear the ringing tone during a call, it is typically triggered by a 180 response. This tells the caller's device to play the local ringback sound while waiting for the other party to answer.
Related: 183 Session Progress · 200 OK · PRACK
183 Session Progress
A SIP provisional response used to deliver Early Media. Unlike 180 Ringing (which triggers a local ringback tone), 183 carries actual audio from the network, such as a recorded announcement or a custom ringback. The response includes SDP so the caller's device knows where to receive the audio stream.
Related: Early Media · 180 Ringing · SDP · PRACK
200 OK
The standard SIP success response. When the called party answers a call, their device sends a 200 OK back to the caller. For REGISTER requests, 200 OK means the registration was accepted. For BYE, it confirms the call has ended. 200 OK is the response you want to see; it means the request was completed successfully.
Related: ACK · INVITE · BYE
302 Moved Temporarily
A SIP redirect response telling the sender to try a different address. The response includes a Contact header with the new address. The sender then re-sends the request to the new destination. This is used by SIP Redirect Servers and for call forwarding scenarios where the system redirects the caller without staying in the call path.
Related: SIP Redirect Server · Call Forwarding
401 / 407 Authentication Required
SIP responses that challenge the sender to provide credentials. A 401 is sent by a server (e.g., a registrar), while a 407 is sent by a proxy. The sender must re-transmit the request with a valid username and password. This is the standard way SIP systems verify that devices and users are authorized.
Related: Credential Authentication · REGISTER · Digest Authentication
403 Forbidden
A SIP response indicating that the server understood the request but refuses to fulfil it. Unlike 401/407, providing credentials will not help. Common causes include dialling a blocked number, exceeding account limits, or lacking permission to use a particular service. The caller must address the restriction before trying again.
Related: 401/407 Authentication Required · ACL
404 Not Found
A SIP response meaning the server cannot find the user or number being called. The SIP URI in the request does not match any registered user. This can happen when someone dials a wrong number, when the called user has not registered their device, or when the domain in the SIP URI is incorrect.
Related: SIP URI · SIP Registrar · REGISTER
408 Request Timeout
A SIP response indicating that the server did not receive a response in time. The destination device may be offline, unreachable, or experiencing network problems. The call attempt is abandoned after the timeout period. Users typically hear silence or a recorded message saying the party is unavailable.
Related: 480 Temporarily Unavailable · OPTIONS
480 Temporarily Unavailable
A SIP response indicating that the called party is reachable but currently unable to take the call. This could mean the user has Do Not Disturb enabled, the device is in an error state, or the user is temporarily away. Unlike 404 (not found), 480 suggests the user exists but is not available right now.
Related: 486 Busy Here · 408 Request Timeout · DND
486 Busy Here
A SIP response indicating that the called party's device is busy with another call. The caller typically hears a busy tone. The system may then try alternative routing (like forwarding to voicemail) depending on the PBX configuration.
Related: 480 Temporarily Unavailable · Voicemail · SIP Forking
487 Request Terminated
A SIP response confirming that a pending request was cancelled. This is the standard reply to a CANCEL message. When you hang up before the other party answers, your device sends CANCEL, and the server responds with 487 to confirm the call attempt has been stopped.
Related: CANCEL · INVITE
488 Not Acceptable Here
A SIP response indicating that the server can handle the request method but cannot accept the media parameters offered. Typically this means the two sides cannot agree on a common codec or media format. The caller may retry with different SDP settings or the administrator may need to configure compatible codecs.
Related: SDP · Codec · INVITE
500 Server Internal Error
A SIP response indicating that the server encountered an unexpected problem and could not complete the request. This is a generic error, similar to a "something went wrong" message. The issue is on the server side, not the caller's side. Retrying after a short wait may succeed if the problem is temporary.
Related: 503 Service Unavailable · SIP Proxy
503 Service Unavailable
A SIP response indicating that the server is temporarily unable to handle the request due to overload or maintenance. Unlike 500, this explicitly signals a temporary condition. Callers or proxies that receive a 503 should try an alternative server or retry later. Properly configured systems use 503 to trigger failover to backup routes.
Related: 500 Server Internal Error · Failover · Trunk Group
Related Sections
📡 SIP Trunking — How SIP connects your PBX to the phone network
📞 Core Concepts — PSTN, VoIP, DID, and foundational telephony terms
🎵 Audio, Media and Codecs — Codecs, jitter, latency, and RTP
🔒 Security — TLS, SRTP, digest authentication, and fraud prevention
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