SIP, The 'quick Quiz'

The correct answer is 5060. SIP (Session Initiation Protocol) is a communication protocol used for initiating and managing real-time sessions such as voice and video calls. The default port for SIP messaging is 5060. This port is used for transmitting SIP messages between devices or applications that are part of a SIP session.

A SIP device needs to send SDP elements in order to negotiate the codec to use with the 'Called' SIP device. SDP (Session Description Protocol) is used to describe multimedia sessions and negotiate the parameters for the session, including the codec to be used for audio or video communication. By sending SDP elements, the SIP device can inform the 'Called' device about its capabilities and preferences for codecs, allowing both devices to agree on a compatible codec for the communication. This negotiation ensures that both devices can effectively communicate and understand each other's audio or video data.

The given statement is false. RTP stands for the 'Real-time Transport Protocol', not the 'Reliable Transport Protocol'. RTP is a network protocol used for delivering audio and video over IP networks. It is designed to provide real-time transmission of multimedia data and does not guarantee reliability or error correction.

DTMF signaling messages can be carried within a SIP INFO message. This means that Dual-tone Multi-Frequency (DTMF) signals, which are used to transmit digits during a phone call, can be included within a Session Initiation Protocol (SIP) INFO message. This allows for the transmission of DTMF tones during a SIP session, enabling functions such as interactive voice response systems or automated attendants to be controlled. Therefore, the statement "DTMF signaling messages can be carried within a SIP INFO message" is true.

The SIP (Session Initiation Protocol) is a signaling protocol used for initiating, modifying, and terminating real-time sessions that involve video, voice, messaging, and other communications applications and services. It operates at the application layer of the TCP/IP protocol stack. This layer is responsible for providing network services to application programs and is where protocols like HTTP, FTP, SMTP, and SIP are implemented. Therefore, the correct answer is Application.

The answer "You decide" suggests that the number of SIP trunks that can be purchased from an ITSP is not predetermined and can vary based on the needs and preferences of the customer. It implies that the customer has the flexibility to choose the number of SIP trunks they require, rather than being limited to specific block sizes.

A Gateway server is required to allow SIP calls to be connected to the PSTN. A Gateway acts as an interface between the SIP network and the traditional PSTN network. It converts the SIP protocol used in VoIP calls into the traditional signaling protocols used by the PSTN, such as ISDN or SS7. This allows SIP calls to be routed and connected to regular telephone lines, enabling communication between SIP-based VoIP networks and traditional telephone networks.

This statement is false. 1 x SIP Trunk does not equate to 1 x DSL 'line' to the ITSP (Internet Telephony Service Provider). SIP Trunking is a technology that allows businesses to make and receive calls over the internet using an IP (Internet Protocol) connection. On the other hand, DSL (Digital Subscriber Line) is a type of internet connection that uses existing copper telephone lines. These two technologies are different and cannot be equated in terms of their functionality and purpose.

When you type an incorrect login or password into your SIP client configuration, you would receive a SIP response code of 401 from the Registrar. This response code indicates that the request requires user authentication. In other words, the Registrar is informing you that your login or password is incorrect and you need to provide valid credentials in order to access the requested resource.

When you move your SIP device from your home to your office, it will have to re-register with the REGISTRAR server. This is because the REGISTRAR server needs to update its records with the new IP address or location of the device. DNS (Domain Name System) is used to translate domain names to IP addresses, but it does not handle the registration process for SIP devices. Therefore, the statement that the REGISTRAR server will be able to locate the device using DNS is false.

SRTP, or Secure Real-time Transport Protocol, does not require a PKI (Public Key Infrastructure) in order to work. SRTP provides encryption, message authentication, and integrity for real-time media such as voice and video over IP networks. While PKI can be used to establish and manage digital certificates for secure communication, it is not a mandatory requirement for SRTP. SRTP can also work with other key management mechanisms, such as pre-shared keys, which do not rely on a PKI. Therefore, the correct answer is False.

A 'STUN' server assists the SIP UA in finding the correct details to put into the SIP message SDP Body. The details that are provided by the 'STUN' server are the external port and the external IP address. These details are necessary for establishing communication between the SIP UA and other devices on the network. The external port is the port number assigned to the SIP UA by the NAT device, and the external IP address is the public IP address of the SIP UA as seen by devices outside of the local network.