8. VOIP 2 Flashcards
What is VoIP?
VoIP is not a protocol but a group of technologies that are used to transport voice info using internet protocol.
Name 3 technologies that are used in VoIP
H.323
Media Gateway Control Protocol (MGCP)
Session Initiaion Protocol (SIP)
What is H323?
H.323 is a communication protocol standard developed by the International Telecommunication Union (ITU-T). It is designed for multimedia (audio / visual) communication over packet-switched networks. Purpose: H.323 enables real-time audio, video, and data communication, making it a key protocol for applications like Voice over IP (VoIP) and video conferencing.
First VoIP Standard to Adopt IETF Real-time Transport Protocol (RTP) - which was transfer of audio/visual over IP network.
A gateway is used to connect to different networks and provides a connection between H323 network and another network.
What is an H323 Gatekeeper?
AN H232 Gatekeeper is a device used on a VoIP network.
It enables address translation and network access control for H323 endpoints.
It utilises the concept of a zone.
A zone is a logical collection of H323 nodes such as gateways and terminals.
There is ONE active gatekeeper per zone.
What are the functions of an H323 Gatekeeper?
The functions of an H323 gatekeeper are:
- Address translation (whereby H323 ids are translated to endpoint IP addresses)
- Call admission control (which entails endpoint registration into the H323 network by allowing or rejecting registration requests.
- bandwidth control (manages endpoint bandwidth requirements)
- zone management (for all registered endpoints wikthin a zone which includes the registration process and alternate destinations)
What is an H323 Gateway?
An H323 gateway connects enterprise VoIP networks to othjer traditional telophony systems including PBX and public land and mobile networks (PLMN).
These gateways can also connect to other parts of the enterprise enterprise such as remote branches, international locations, mobile units.
An H323 gateway can support a number of different signalling protocols including H323, MGCP and SIP
Generally serves as a Digital Signal Processor (DSP). that facilitates media transcoding, conferencing & PCM conversion of voice signals between IP telophiny and the PSTN.
What is an MCU (Multipoint Control Unit)?
An MCU is a server-based device or software that facilitates communication in multi-party audio or video conferencing.
The MCU acts as a central hub, receiving audio and video streams from all participants in a conference.
It processes these streams by mixing or transcoding them into a single stream.
The resulting stream is then sent back to all participants, ensuring everyone can hear and see each other.
This approach reduces the processing load on individual devices, making it easier for participants with limited bandwidth or processing power to join large-scale conferences. However, it can be resource-intensive for the server hosting the MCU
What is the Media Gateway Control Protocol (MGCP)?
AKA H.248 or Megaco
ISpecified inn ETF RFC 2705
It is a Plaintext protocol
It is Used by calling devices to manage IP telophony gateways
Has advantage of centralised gateway administration - allows for scaleable IP telophony solutions
What is SIP (Session Initiation Protocol)?
SIP is:
- defined by IETF RFC 3261
- a P2P Protoocol whereby call routing and session management are distributaed across nodes.
- the protocol is comprised of initiation, modification & termination of interactive multimedia sessions.
- used for audio visual conferencing, chat sessions, game sessions, file trasnfer. & fax over IP.
- SIP does not encode audio info in phone call and does not transport audio info. SIP initiates, terminates communication sessions and the session can be a voice call or a video conference call between 1 or more people.
- SIP can be used with a number of technologies incl VoIP. SIP can be used to SET UP and TERMINATE the call
- SIP sends messages between end points on the internet which are known as SIP addresses.
- SIP bassed telophony networks often use the call processing features of SS7 - Signalling system 7 (SS7)
- Similar to H323, a SIP network will use a gateway to connect to gtraditional telophony network
What are the differences between H323 and SIP?
H323 is based on telophony. SIP is based on use of the internet.
H323 is recommended by ITU, whereas SIP is designed by the IETF
H323 uses an alias - a gatekeeper map which is the host or a telephone number. SIP uses a url
H323 is an application protoocol used for VoIP incl video and audio conferencing, but is NOT used for gaming or file transfer. so the uses of H323 are limited compared to SIP.
SIP is compatible with H323.
What are the advantages of H323? Why use it is it’s uses are limited compared to DSIP
- SIP has Greater Interoperability. Because of it’s inbuilt features that deal with network connections and device failure
- Bandwidth - performs better in terms of bandwidth because it uses a Binary Coding System compared to SIP which is text based (encoded with ASCII)
- H323 has Load Balancing capabilities that SIP does not
- Maintains a flexible method for addressing not present in SIP. Can also Support Multiple Addressing Schemes
What are the main 4 network problems with VoIP?
- Delay / latency
- Jitter
- Packet loss
- Voice activity detaction (VAD)
What are the main types of latency or delay problems with VoIP?
3 types of delay in packet based voice networks:
- propagation delay
- processing delay
- serialisation delay aka queing delay.
What is Propagation delay?
Propagation delay is delay caused by the physical distance the signal must cover and the medium through which it travels - e.g., fiber optic cables or copper wires)
What is processing delay?
Processing delay is - the time taken by network devices (like routers, switches, and the sender/receiver’s hardware) to analyze and process the packets of data. This delay occurs at various stages of a packet’s journey, including:
- Packetization,
- Compression/Decompression,
- Routing/Switching (also refererd to accumulation delay as the packets accumulate in a buffer before they are released).
What is serialisation delay (aka queueing delay)?
Serialisation delay aka queing delay is the time it takes to place a voice packet into an interface. Packets are sometimes held in a queue due to congestion on an outbound interface. meaning more opackets are sent out than the interface can handle in any given moment.
What is jitter?
Jitter refers to the variation in the time it takes for packets to travel from the sender to the receiver.i.e the difference between the expected time and the time of the actual reciept
Ideally, packets in a stream should arrive at regular intervals; however, due to network congestion, route changes, or hardware issues, the arrival times can fluctuate.
Why is jitter important in VoIP?
In voice or video communication, consistent packet delivery is crucial to maintain smooth and natural playback.
High jitter can cause packets to arrive out of order or at uneven intervals, leading to issues like choppy audio, delays, or dropped words.
How is jitter handled?
VoIP endpoints (Devices) often use a JITTER BUFFER to temporarily store and reorder packets, ensuring they are played back at a consistent pace. Some jitter buffers can be adjusted to compensate for delays, so the more jitter you have, the larger the jitter buffer needs to be.
Many vendors use STATIC jitter buffers that do not adjust. Cisco uses dynamic jitter buffers. Dynamic is preferable in producing superior call quality.
A well-managed network with sufficient bandwidth and minimal congestion helps reduce jitter.
How can packet loss be managed?
Packet loss is not a massive problem as long as the dropped packets are not percievable to the human ear.
One way to ensure relaibilty is to use Quality of Service to make packets resistant to packet loss
Cisco has developed many QoS tools to classify and manage traffic. including Cisco VoIP implementation enables a voice configured browser to respond to intermittant packet loss - i.e. if a voice packet is not recieved it is assumed to be lost so the last packet is replayed.
What is Voice Activity Detection (VAD)?
At least 50% of bandwidth is wasted ina conversation while the other party is speaking, and gaps in sentences.
VAD utilises this wasted bandwidth for other purposes. The speech is detected in decibels and when no speech, voice data is not transmitted.
Therefore VAD detects a drop off of speech amplitude and waits a fixed amount of time before creating and inserting speach frames into packets
What are the problems with VAD?
- background noise (not important in convo but may be transmitted - it might not be able to distinguish between speech and background noise. Called - signal to noise threshold
- front end speech clipping caused by the detection of when someone starts to talk being detected too slowly.
List some voice quality problems
Noise:
Absolute Silence (when you can’t tell if someone is still there - no sound)
Clicking (external sound often inserted at intervals caused by clock slips or digital errors on the line
Crackling (a form of light static - caused by electrical interference or poor electrical connection)
Crosstalk (where you hear someone else conversation - caused if cables are too close together)
Hissing (more constant than static can be caused by VAD)-
Static
Voice Distortion:
Echoed Voice
Garbled Voice
Volume Distortion
