Yes, there should and must be a media mixer in the conference session. The exert essentially receives ART streams from one or more sources, combines them in some way and forwards the new mixed stream to one or more receivers. D. Timestamp to timetable conversion : Ran-Ri /GHz z) (2100-980)/8000 = pieces Therefore, packets of size 140 sec have been picketed. The beginning of talkers is indicated by the marker bit. It also helps adjust the pullout delay at the receiver and is also used for silence suppressions to indicate packets that follow periods of silence. F.
Sequence number in the ART header helps reorder the ART packets at the receiver end. And the timestamp helps play out the voice packets in sequence. . The payload type field in the ART header indicates the payload type and size. If it’s PC that’s used for digitization, then the payload type indicates the same. The sampling frequency being kHz in case of PC the sample size or payload size will be KICK = 15. 6 sec per sample. H. The sequence number in the ART P header helps detect the R TAP packet loss 2. Consider voice conference session with 28 participants that each simultaneously sends and receives 102 Kbps ART streams.
Determine ROTC packet transmission period for each sender and receiver assuming average ROTC packet size for the session to be 128 Bytes. Round your result to the nearest hundredths. (18 points) Destined t] # of senders * aver. _ R TCP _ packet_ size 0. 05*0. 25* session bandwidth = (28*1 = mass Tree shiver D # of receivers * aver. _ ROTC _ packet_ size 0. 05*0. 75* session _ bandwidth = mass 3. Draw and explain five common Poi connection strategies we covered during the first lecture that transport voice over IP in at least one segment of the communication network.
Make sure you include all required components to make interconnectivity between them possible. How Poi systems communicate with each other and T DIM legacy POTS systems. (18 points) Poi Connection Strategies Strategy 1: Poi Base to Poi Base over IP Network Strategy 2: Poi Base to Poi Base over Voice Network : POI Base to POTS Base over IP Network Strategy 3 Strategy 4 : Poi Base to POTS Base over Voice Network Strategy 5: POTS Base to POTS Base over IP Network Strategy 6: POTS Base to POTS Base over EST. 4. . Compare following set of Codes. Include algorithms they use and the efficiency of those, bandwidth requirements, delay impact, packet loss handling, BAD etc. In what specific situations would you use each of them? Explain your reasoning. G. 711 G. 726 G. 728 G. 723. 1 G. 729 G. 711: C] Most common today A waveform code If uniform quantization were used, it would take 12 bits/sample 96 Kbps; uniform quantization gets the same results with 8 bits/sample a. K. A.
Pulse Code Modulation (PC) C] The best quality conventional-band (narrowed ) code C] The workhorse of the EST. for digital trucking and switching D Foundation for DOS – 64 Kbps TWO PC variants 0 Um-law – Used in North America and Japan, slightly skewed to be “friendlier” to lower signal levels C] A-law – used in Europe and the rest of the world and international routes Both provide good quality C] PC sends each sample independently across the network Does not support BAD G. 26: C] ADAPT waveform type code C] 32 Kbps rate is commonly used for compression in TDMA networks, private networks, undersea cables, and satellite links G. 726 rate: 32 Kbps = (2 x 4 kHz) x 4 bits/sample C] G. 726 rate: 24 Kbps = (2 x 4 kHz) x 3 bits/sample C] G. 726 rate: 16 Kbps = (2 x 4 kHz) x 2 bits/sample C] More sensitive to data loss than G. 711 because the decoder can lose its adaptive reference, and it takes a finite time to reconvert 0 Low-delay C] Does not support BAD G. 728: G. 728 – approved in 1992 OLD-CELL hybrid type code Operating at Kbps Allows network signaling to go through
Law-delay Does not support BAD GAGA. 1 C] CELL hybrid type code 0 VERY low bandwidth Has two modes of operation 6. Kbps or 5. Kbps that can change dynamically at each frame (both of them mandatory for implementation) supports BAD, TXT and CNN C] Disadvantage: introduces delay of apron. 37. 5 ms; it doesn’t transmit TDMA tones and fax G. 729 : CSS-CELL hybrid type code Operates at 8 Kbps Introduces delay of apron. 15 ms Doesn’t transmit TDMA tones and fax 0 Same frame structure C] Less computation -> slightly lower quality G. 729.
B Supports BAD Based on analysis of several parameters of the input The current frames plus two preceding frames TXT Send nothing or send an SIDE frame C] SIDE frame contains information to generate comfort noise b. Is it possible to change CODES during the call? How many R TAP streams (flows) you need to establish full-duplex voices conversation? (1 6 points) Yes, codes can be changed or negotiated during the call. The processing of encoded voice to/ from ART/IP packets and sending/receiving these packets across the network can require a substantial amount of CPU power.
Audio encoding mechanisms such as GAGA. 1 encode moms samples of voice into a packet. This results in a Seibel packet rate of 33. 3 posts/sec in a normal half-duplex conversation (1 talker, 1 listener). In a full duplex situation (both parties talking at the same time) one conversation can have a peak of 66. 6 posts/sec. A Gateway system of just 1 00 users could therefore have a possible peak of 6,666 packets per second. 5. What is the delay budget for Poi application and why it is important? What delay components need to be considered as part of the delay budget calculation?
Briefly elaborate each of them. (16 points) Delay Budget The end-to-end delay in a Poi network is known as the delay budget. Network administrators must design a network to operate within an acceptable delay budget. Delay is the accumulated latency of end-to-end voice traffic in a Poi network. The purpose of a delay budget is to ensure that the voice network does not exceed accepted limits of delay for voice telephony conversation. The delay budget is the sum of all the delays, fixed and variable, that are found in the network along the audio path.
You can measure the delay budget by adding up all of the individual contributing components, as shown in the figure. The delay budget is measured in each direction individually, not round-trip. Network administrators must be aware that delay exists, and then design their network to bring end-to-end delay within acceptable limits. Need For a Delay Budget As delay increases, talkers and listeners become unsynchronized and often find themselves speaking at the same time or both waiting for the other to speak. This condition is commonly called talker overlap.
While the overall voice quality may be acceptable, users may find the stilted nature of the conversation unacceptably annoying. Talker overlap may be observed on international telephone calls that travel over satellite connections. Satellite delay is about 500 s: 250 ms up and 250 ms down. Following is an explanation of six major factors that contribute to overall fixed and variable delay: Coder delay: Also called processing delay, coder delay is the time taken by the ADS to compress a block of pulse code modulation (PC) samples.
Because different coders work in different ways, this delay varies with the voice coder that is used and the processor speed. Factorization delay: Factorization delay is the time it takes to fill a packet payload with encoded or compressed speech. This delay is a function of the sample block size that is required by the vectored and the number of blocks placed in a single Ramee. Factorization delay is also called accumulation delay because the voice samples accumulate in a buffer before being released. With typical payload sizes used on Cisco routers, factorization delay for G. 11, G. 726, and G. 729 does not exceed 30 ms. Queuing delay: After the network builds a compressed voice payload, it adds a header and queues for transmission on the network connection. Because voice should have absolute priority in the router or gateway, a voice frame must wait only for a data frame already playing out or for other voice frames ahead of it. Essentially, the voice frame waits for the serialization delay of any preceding frames that are in the output queue. Queuing delay is a variable delay and is dependent on the trunk speed and the state of the queue.
Serialization delay: Serialization delay is the fixed delay that is required to clock a voice or data frame onto the network interface; it is directly related to the clock rate on the trunk. Network delay: The public Frame Relay or ATM network that interconnects the endpoint locations is the source of the longest voice-connection delays. These delays are also the most difficult to quantify. If a private enterprise builds its own internal Frame Relay network for the purpose of wide-area connectivity, it is possible to identify the individual components of delay.
In general, the fixed components are from propagation delays on the trunks within the network; variable delays are the result of queuing delays that clock frames into and out of intermediate switches. To estimate propagation delay, a popular estimate of 10 microseconds/mile or 6 microseconds/km (G. 114) is widely used, although intermediate multiplexing equipment, backhanding, microwave links, and other features of carrier networks create many exceptions. Typical carrier delays for U. S.
Frame Relay connections are 40 ms fixed, and 25 ms variable, for a total worst-case delay of 65 ms. Dejected buffer delay: Because speech is a constant bit-rate service, the jitter from all the variable delays must be removed before the signal leaves the network. In Cisco routers and gateways, this is accomplished with a dejected buffer at the far-end (receiving) router or gateway. The dejected buffer transforms the variable delay into a fixed delay by holding the first sample that is received for a period of time before playing it out.
This holding period is known as the initial pullout delay. The actual contribution of the dejected buffer to delay is the initial pullout delay of the dejected buffer plus the actual amount of delay of the first packet that was buffered in the network. The worst case would be twice the dejected buffer initial delay (assuming the first packet through the network experienced only minimal buffering delay). 6. How define Poi in the realm of Real Time Communication? What causes the disruption in packet timing and how to manage voice interstates synchronization? 16) Real-time communications (ROTC) is any mode of telecommunications in which all users can exchange information instantly or tit negligible latency. In this context, the term “realties” is synonymous with “live. ” R ETC can take place in half-duplex or full-duplex modes. In half-duplex ROTC, data can be transmitted in both directions on a single carrier or circuit but not at the same time. In full-duplex ROTC, data can be transmitted in both directions simultaneously on a single carrier or circuit. ROTC generally refers to peer-to-peer communications, not broadcast or multicast.
In ROTC, there is always a direct path between the source and the destination. Although the link might contain several intermediate nodes, the data goes from resource to destination without having to be stored anyplace. In contrast, timesharing communications always involves some form of data storage between the source and the destination. No end-to-end protocol, including ART P, can ensure in-time delivery. This always requires the support of lower layers that actually have control over resources in switches and routers.
ART P provides functionality suited for carrying real-time content, e. G. , a timestamp and control mechanisms for synchronizing different streams with timing properties. Disruption in packet timing:The need for an ordered synchronous sample flow s quite essential in voice communication. Unless common clock terminals have independent clocks, without timing information receiver would not know when to play a received frame (each packet doesn’t necessarily include same amount of voice and they are not always sent periodically).
Jitter causes the disorder. It’s because of jitter there is a difference in inter-arrival times and thus it leads to packet timing disruption. To counter this play out buffer is introduced. While some algorithms simply adjust the buffering delay of received packets in order to maintain a safe level of buffer occupancy most rely on timing information in the Oromo of timestamps (ART) – some codes contain timing (MPEG) Voice Interstates Synchronization:ROTC helps manage the voice interstates synchronization. L R TCP can provide absolute Walcott timestamps that allow the receiver to map actual time to the sender’s C] ART timestamps. ROTC can also optionally provide minimal session control information. C] ROTC packets are sent periodically, with the sending interval dependent on the size of the C] session (number of participants) and bandwidth limitations. A minimum interval between 0 ROTC packets of 5 seconds is suggested. Each ART flow has an associated control flow that provides:
C] Time-based management and information for synchronization Feedback on the quality of data distribution C] Calculating round-trip time 0 Sent and lost packets Calculating jitter C] Timing of ROTC packets C] Member identification and management Each participant in ART P session periodically transmits R TCP control packets to all other participants Feedback can be used to control performance: Sender may modify its transmissions based on feedback The specification doesn’t dictate what the application should do with this feedback information; this is up to the application developer.