This article contains common terminology that you will encounter when working with SIP for VoIP. If you are new to SIP, this is a great starting point or reference tool for engaging with other articles. There is also added context to help better understand the SIP.US platform.
NOTE: You will also find Number Porting and Messaging sections in this Knowledge Base with corresponding articles that provide a glossary of terms on each topic.
SIP – (Session Initiation Protocol) is a signaling protocol used for telephony over the internet. It is responsible for initiating, maintaining, and termination communications sessions. All of SIP.US voice services utilize SIP signaling and adhere to industry standards.
VoIP – (Voice over Internet Protocol) basically refers to telephony over the internet. It refers to the transmission of voice as data packets, typically in UDP format for the best end-user experience. It integrates with PSTNs and the wider telecommunications world while offering enhanced features over traditional POTS.
PSTN - (Public Switched Telephone Networks) makes up all of the world’s interconnected telephone networks. It consists of decades of technological advancements in physical and digital infrastructure with unifying standards that allow interconnections of different technologies new and old.
POTS - (Plain Old Telephone Service) is the traditional analog signal telephone service. POTS was eventually adapted to ISDN digital networks that operate on the same physical cabling infrastructure. POTS is still in service in certain locales, oftentimes remote locations, but is still able to connect to modern voice telephony such as VoIP (with some limitations).
NPA - (North American Numbering Plan) refers to the standardized format for North American telephone numbers. The NPA further defines area codes based on the geographic regions. The NPA applies to dialing patterns, and is 10 digits in full including the three-digit area code followed by the three-digit prefix, and lastly the four-digit exchange. An example of an NPA would be: 800 250 6510 (note this is the SIP.US TF number).
SIP Trunk – This is a common identifier in SIP which is used to identify a user ID and associate it to an account and billing plan. SIP trunks are commonly used for the input value in username, alias, authentication ID fields, etc. in phone systems. A SIP trunk is retrieved from the customer portal for the SIP credentials when registering a phone system to the SIP.US gateways.
SIP Registration – Once you have a SIP trunk with an active billing plan, you can send a REGISTER request from your phone system to the SIP.US gateways to establish the connection for the flow of call traffic. The SIP registration requires the SIP credentials along with IP information for where to direct the traffic. We then display the active registrations and user agent details for your SIP trunks in the customer portal.
User Agent – This refers to the the type of phone system you register with your SIP trunk. The phone system will transmit its information (model and version) in the user agent line of the SIP REGISTER packet.
SIP Credentials – These make up both the SIP trunk number and the password for that SIP trunk. You can retrieve this information directly from the customer portal in the SIP Trunking tab.
IP Authentication – Besides SIP Credentials, customers may also choose to assign a public static IP address to their SIP trunk in the customer portal. All inbound traffic will be sent to that IP address, and all outbound traffic passing from it will be allowed. The CONTACT header in the SIP messaging will require the corresponding IP in order to work.
DID - (Direct Inward Dialing) refers to a phone number with a unique terminating destination. SIP.US sells DIDs which can then be associated to a SIP trunk and programmed to receive calls on your phone system. When we receive the traffic, we will route it through the associated SIP trunk and to the phone system registered with that SIP trunk.
CDRs – (Call Detail Records) are the phone records associated with your account. These contain information such as the timestamps, calling number, called number, the SIP trunk the traffic routed through, the call result, and call duration. SIP.US offers (automated) historical call records via the lookup function for usage reports/audits via the customer portal.
PBX - (Public Branch Exchange) is a common phone system platform. There are different types of phone systems, but PBX systems are common in VoIP as accessible solutions for setup, affordability, and scalability while still offering enough customization options for a wide range of deployment requirements.
CNAM – (Caller ID Name) is a feature of the US public telephone network that associates a text name with a Caller ID (phone number). It allows users the benefits of associating their personal name or business with a phone number which can lead to increased recognition by recipients, fewer unanswered calls, and prevent calls from being flagged as spam. CNAM is an optional feature, meaning not all carriers support it or may charge to support it. When a call is received, the receiving end performs a lookup on the Caller ID which is then associated with the CNAM on record. CNAM is never transmitted from the calling (originating) party in the US telephone network.
Caller ID – This refers to the numerical string transmitted as the calling number, or number called from. When a SIP INVITE is passed, a caller ID is required in the FROM header. Customers may send whatever called ID they like, but we enforce STIR/SHAKEN and call signing attestation. Customers are also responsible for the caller ID they transmit in regards to legitimacy, as we enforce a zero-tolerance fraud policy.
DNS - (Domain Name System) provides a naming system for IP addresses in which an end user can query the DNS with a domain name and the system will resolve that domain name to a unique IP address. For example, gw4.sip.us resolves to (104.219.162.40). Some phone systems may specify a domain name in the registration and will resolve the registration request to the corresponding IP address, while others may accept direct input of an IP address for the host/proxy/SIP server.
SRV - (Service Record) as it pertains to SIP is a domain that the DNS can query and it will resolve with a hierarchy of weighted IP addresses. For example, a PBX would be configured with the SRV gw.sip.us. In the event that gw4.sip.us is unreachable, it would resolve to the next weighted IP address gw5.sip.us and so on.
NAT - (Network Address Translation) is a method for rewriting IP addresses in the IP headers of packets as the packets traverse through a network with the goal of reaching a different IP address such as a Private IP destination.
Public IP - The IP address assigned by an ISP that allows a network to communicate publicly with other networks through the internet. A public IP address is unique and is therefore able to be defined to a specific location much like a street address (however, public IPs are not always Static). In VOIP, the public IP is populated in the contact address of the SIP traffic and that tells our servers where to direct your registered inbound traffic and audio, or, how to authenticate your outbound traffic as another example.
Static IP – A public IP address that never changes. ISPs often assign Dynamic IP addresses, which can change over time after modem resets, power outages, etc. Clients can request/pay for a Static IP address for consistency which allows for security options in VoIP such as IP authentication for all traffic and other controls when building out advanced networks.
Private IP – The IP address assigned by the local network to a specific device. Private IPs populate a specific range that only a local network can deliver traffic directly to. Private IPs cannot be used in VOIP without a NAT policy that displays the public IP address in the contact header, because SIP.US cannot send traffic to a private IP since it only exists in the local network.
Rate Plan – This defines the billing plan ordered and assigned to the SIP trunk. Customers may choose between channel or minutes package options based upon what suits their needs. Our Billing and Sales teams are available to assist you with selecting a Rate Plan.
Term Rate Plan – Not to be confused with a Rate Plan, the Term Rate Plan defines the destinations the SIP trunk is able to send outbound calls to. Our default Term Rate Plans are limited to US48 and Canada, but different plans can be enabled on a SIP trunk based on the user requirements. See Setting Up Outbound Calling for more information.
Channel – SIP.US offers an unlimited monthly calling package for calls to the US48 and Canada and bills based off how many channels, or lines, the customer purchases. Each channel allows for an active call path, and simultaneous active calls will occupy the total channel capacity purchased on a SIP trunk. Therefore, users can place as many calls as they want, as long as they do not occupy all of the channel capacity. Once a user attempts to exceed the purchased channel capacity, those calls will fail. The user would either have to wait for other calls to end and reopen the channel capacity, or purchase more channel capacity.
Dialing Pattern – This refers to the string of numbers dialed for outbound calls. The standard US dialing pattern follows the NPANXXXXXX format. SIP.US requires you to format your dialing patterns to successfully complete outbound calls. More information can be found in this article on Setting Up Outbound Calling.
DTMF - (Dual-Tone Multi-Frequency) provides input feedback in the form of audio tones when a user presses an input on their phone dialer. For example, an IVR may provide department options (press 1-4) and will confirm the user’s input with a tone when they select one of the department options. DTMF provides a response confirmation that the input was successfully transmitted. SIP.US supports DTMF tones.
IVR - (Interactive Voice Response) is a system that allows telephone users to interact with an automated system to complete actions. IVRs answer calls and issue voice prompts, which users can respond to via numerical input or other modes of response. Advanced IVRs allow you to speak voice commands and perform other actions depending on the capabilities of the systems for accomplishing tasks such as account management, completing call transfers, or providing informational services.
ITSP - (Internet Service Provider) refers to the internet provider the client uses to reach SIP.US services.
Softphone – A software program that processes telephone calls over the internet. Softphone apps are available on multiple platforms including desktop computers, laptops, smart phones, tablets, and other devices. These programs are designed to work much like a standard calling experience with a desk phone or cell phone, but typically include a suite of configurable options depending on the developer. SIP.US is compatible with a number of softphone apps, and setup works similarly to a typical PBX. Softphones tend to cater towards individual users or small groups of users versus PBXs which are designed with scalability in mind for organizations.
Status Codes – You will encounter SIP messaging whenever you observe a call flow or packet capture of a call. Note the following explanations of what action is taking place in correspondence with each status code.
- INVITE – For inbound, the call invitation delivered from SIP.US to the registered phone system and destination point (user). It contains the destination number and the number called from. For outbound, the call invitation originated from the registered phone system. It must contain a legitimate, defined caller ID and the destination number as well as matching SIP credentials to authenticate the traffic.
- 100 TRYING – For inbound, the phone system is processing the inbound call delivered from SIP.US. For outbound, SIP.US is attempting to reach the destination point.
- 180 RINGING – For inbound and outbound, the phone system is ringing the endpoint (user).
- 200 OK – For inbound, the phone system has accepted and established the call and sends confirmation. For outbound, the far end has accepted the established the call.
- ACK – For inbound, the 200 OK reaches SIP.US from the phone system and SIP.US understands the call is now active and acknowledges. For outbound, the 200 OK from the far end reaches SIPTRUNK and the acknowledgement is sent back to the far end.
- 486 BUSY – For inbound, the phone system reports that the endpoint (user) has received and rejected the call invite. This could be due to manual rejection, Do Not Disturb, or improperly configured user settings. For outbound, the far end rejects the call. This could be a manual rejection, a carrier rejection (spam suspected), the receiving party has blocked the caller ID, or the receiving party is on another call and the line is busy.
- BYE – For inbound, the phone system sends a request to terminate the call to SIP.US. For outbound, the request comes from the upstream pier. It can either indicate the far end (receiving party) ended the call, or the call dropped at some point in the network.
SIP Message Headers- All SIP messages contain headers, but the most common packets that will require troubleshooting are the REGISTER and INVITE messages. The following descriptions provide basic definitions of the most common headers we find user configuration issues with. Refer to specific troubleshooting guides for more information.
- FROM - REGISTER packets must contain the SIP trunk number and the SIP.US gateway host. INVITE packets must contain the caller ID and the SIPTRUNK gateway host.
- TO - REGISTER packets must contain the SIP trunk number and the SIP.US gateway host. INVITE packets must contain the destination number and the SIPTRUNK gateway host.
- AUTH - This header contains the host/realm (the SIP.US gateway host), username (SIP trunk number), and the security credentials (password).
- CONTACT – REGISTER and INVITE packets must contain the client’s public IP address and port number SIP.US should contact the equipment at.
RTP - (Real-time Transport Protocol) is the network protocol for delivering audio and video over IP networks. RTP runs over UDP, but is differentiated in SIP from UDP. SIP.US delivers call audio via RTP by default. Typically, you will need to define a separate listening port range for RTP in your network and PBX, separate from UDP.
UDP - (User Datagram Protocol) as it relates to telephony refers to the unidirectional flow of data packets from one point to another. It does not guarantee end-to-end transmission of the data in favor of low latency transmission. It is used in telephony, as it is ideal for delivering voice for real-time conversation and considered reliable enough for the use case. SIP.US, by default, utilizes UDP to deliver SIP and audio packets.
TCP - (Transmission Control Protocol) works counter to UDP in that it requires an end-to-end “handshake” to ensure reliable data transmission for ALL packets. This in turn allows for additional voice security options in which traffic can be encrypted, but comes at the penalty of increased transmission complexity and latency.
TLS and SRTP - (Transport Layer Security/Secure Real-time Transport Protocol) allows for encryption of SIP and voice packets for sensitive data transmission. Contact SIP.US Support if you wish to configure TLS-compliant service for your call traffic.
Audio Codec – The encode/decode format for audio data transmission. SIP.US supports G.729 and G.711 u-law and the optional a-law. The audio codec will need to be configured in the user’s phone system.
Direct Media (RTP) - SIP.US utilizes Direct Media delivery in which it releases the call to the direct pier carrier after the call has been established via SIP. This provides a lower latency experience, but requires proper firewall whitelisting rules. See this article on Direct Media Delivery and Missing Audio for more info.
PSTN Forward – An automatic forward that engages whenever an inbound call is received for the destination number. The PSTN Forward ignores any registered phone system and automatically forwards the call back out to the programmed number for receiving the forward. The forward must be defined in the customer portal.
PSTN Backup – A forward that only engages when the registered phone system fails to connect the inbound call to an end user or IVR. The PSTN Backup is ideal for failover. The forwarding number must be defined in the customer portal.
Primary Trunk – All DIDs are assigned to a SIP trunk by default (acts as primary) and DIDs must be assigned to a SIP trunk in order to route to a phone system. The Primary is the first inbound route SIP.US will try when delivering a call INVITE. The SIP trunk must be registered and the phone system must provide a response.
Secondary Trunk – A secondary SIP trunk assignment option is available in the customer portal. In the event that the Primary Trunk routing fails to receive a response, SIP.US will advance the routing to the Secondary Trunk as a failover option. Please note that Secondary Trunks always require manual assignment to DIDs.
Linked Trunk – A generated SIP trunk that’s assigned to an existing billing plan. The Linked Trunk allows flexibility for when customers need more segmentation options for their VoIP deployment. A Linked Trunk shares the minutes package or channels of the billed SIP trunk it is associated with. Linked Trunks use unique credentials and can be used for failover, different site locations, different SIP settings, etc. Linked Trunks may be requested from the Support team for free.