What Do You Know About SIP-VoIP? Trivia Questions Quiz

The voice quality of a VoIP call is directly affected by loss, latency, and jitter. Loss refers to the loss of packets during transmission, which can result in choppy or distorted audio. Latency is the delay in the transmission of voice data, causing a noticeable lag between the speaker and the listener. Jitter refers to variations in the delay of packet arrival, leading to inconsistent voice quality. All of these factors can significantly impact the overall voice quality of a VoIP call.

Jitter refers to the variability in the delay of data packets or signals as they are transmitted over a network. It is a measure of the inconsistency or fluctuations in the timing of the packets. When there is high jitter, the packets arrive at the destination with varying delays, causing issues such as packet loss, data corruption, and poor quality of audio or video streams. Therefore, the correct answer is "Variability of delay."

H.323 is indeed an umbrella standard, meaning that it encompasses multiple different standards within one overall standard. This allows for interoperability and compatibility between various communication devices and systems that adhere to different standards. By incorporating these different standards, H.323 provides a comprehensive framework for audio, video, and data communication over IP networks.

SIP and H.323 are both protocols used in Voice over IP (VoIP) communication. They have similar functionalities such as call control, call setup, call teardown, call waiting, call hold, call transfer, call forwarding, call return, call identification, call park, and capabilities exchange. Therefore, the statement that SIP and H.323 are functionally similar is true.

A device that performs conversion between the VoIP world and traditional telephony is called a gateway. Gateways are used to connect VoIP networks with traditional telephone networks, allowing communication between the two systems. Gateways convert the voice signals from analog to digital or vice versa, enabling seamless communication between VoIP and traditional telephony users.

SIP (Session Initiation Protocol) borrows much of its syntax and semantics from HTTP (Hypertext Transfer Protocol). This is because both protocols are used for communication purposes. HTTP is primarily used for web browsing and transferring hypertext documents, while SIP is used for initiating, modifying, and terminating real-time sessions involving video, voice, and messaging applications. By borrowing from HTTP, SIP is able to leverage existing standards and protocols, making it easier to integrate and communicate with web-based applications and services.

The one-way latency for VoIP calls should be no more than 150 ms. This is because any latency beyond this threshold can result in noticeable delays and disruptions in the conversation. Keeping the latency within this limit ensures a smooth and real-time communication experience for the users involved in the VoIP call.

Establishing communication using SIP typically involves four steps: session initiation, session description, session establishment, and session termination. Therefore, the given statement is false as it only mentions three steps.

G.728 would be the audio codec to consider for voice compression if delay were an issue. This codec is specifically designed to prioritize low-delay speech compression, making it suitable for real-time applications such as voice communication. It achieves this by utilizing a low-complexity algorithm that reduces the coding delay, ensuring minimal latency in the compressed audio signals.

SIP (Session Initiation Protocol) and H.245 are two protocols that can be used for capability negotiation. SIP is a signaling protocol used for initiating, modifying, and terminating communication sessions, such as voice and video calls, over IP networks. It allows devices to negotiate their capabilities and agree on the protocols and codecs to be used for the communication session. H.245, on the other hand, is a protocol used in videoconferencing systems to negotiate capabilities for media channels, including audio and video codecs, bit rates, and resolutions. Together, SIP and H.245 enable devices to negotiate and establish communication sessions with compatible capabilities.