BIOLOGY Paper - Amphibian Development - Human Development Essay

BIOLOGY Paper - Amphibian Development - Human Development - Essay Example s belong to Tetrapods or four-legged vertebrates and are considered as ectothermic vertebrates.3 Physically, frogs’ skin is smooth, has three-chambered heart, and does not have claws on their toes. Frogs usually lay their eggs in water until it hatches to into tadpole or larva. Tadpoles has to be on the water since they breathes with their gills. Chicks are classified as endothermic vertebrates. 3 Their physiological appearance includes skin covered with feathers, has four-chambered heart, lightweight bones that are hollow, and with wings. Similar to frogs, chicks also lay eggs. In comparison between the eggs produced by chicks and frogs, the egg that comes from chicks has more yolk than the frogs. Since human fertilized egg or the zygote does not contain any yolk, we will take into consideration the frog’s reproduction and development as the best choice in comparison with the human reproduction and development process. There is a huge difference between a frog egg (mesolecithal egg) and human ova (egg cells). Mesolecithal egg has a moderate amount of yolk being distributed throughout the cytoplasm of the egg with more concentration towards the ‘vegetal pole.’ On the other hand, the human ova do not show traces of yolk. Also, the external portion of the frog egg has a smooth surface while the human ova has an uneven surface. Frog egg is divided into a dark pigmented hemisphere called the animal hemisphere while the lightly pigmented hemisphere is known as the vegetal hemisphere. The gray crescent is the dorsal surface of the embryo that plays a role in establishing the embryonic axis. See figure 1. Human ova, measures around 0.2 mm. in diameter, are enclosed within the egg follicles of the ovaries.5 The cell substance of the ovum i is called oà ¶plasm.10 (in amphibians, oà ¶plasm is equivalent to the yolk) The nucleus is known as the germinal vesicle while the nucleolus is the germinal spot. The ovum is enclosed by a thick transparent envelope called the

Midterm Exam Employment Law 2009 Research Paper

Midterm Exam Employment Law 2009 - Research Paper Example The company was a large international firm in San Francisco with almost 5000 employees. Jami had given the interview with the employment group of 6 people. One of them was John Clark, a V.P and the others were Human Resources V.P, V.P in charge of sales and two other supervisors. During the interview, she noticed many of the unwanted things i.e. all of them in the selection panel kept a special eye on her physical features and was asked all kinds of personal and absurd questions just before the interview was going to complete. Despite this unfamiliar circumstance, she joined the job with an insurance benefit of 401K plan and annual salary of US$80,000.00. She was also said that after three months of probation period she could be terminated with a valid cause. After her efficient work of six months, she was offered a lunch by Mr. John Clark, V.P. In the dinner, Clark said that the company needs another V.P just like Jami. In addition, he also offered Jami to spend the weekend with him in Yosemite so that the details of the plan can be discussed. When they visited the place together, Jami came to know that Clark was having an eye on her from the very first day and she would be provided with name and fame if she agrees to all the unwanted offers of Clark. When she refused to the offer she had to go through a lot of harassment in the office. Her performance was considered to be poor and she was not allowed a leave for her mother’s surgery. Irrespective of all these adversities, she went to visit her mother without a leave notice. On returning, she was demoted to a supervisor and had a decline in salary of US$40,000.00 per year. After all these occurrences, she lodged a complaint with the Equal Employment Opportunity Commission (EEOC) and therefore sued the company to the Federal Court. It can be viewed that all workplace related issues in the form of cases are handled by the Equal Employment Opportunity Commission (EEOC) of the US (State Human Resources Manu al, â€Å"Title VII of the Civil Rights Act†). According to the case provided, Jami Jensen has been the victim of sexual harassment by the Vice President of the company, wherein she was working. When an offer was provided to Jami and not accepted by her, the V.P. threatened to sue her from the company and also lessened her salary to a huge proportion. The V.P. also kept a condition that in order to avail the facilities of the company, she must take back the case and in return, she would be rewarded with a higher position in the company and a greater hike in her salary. However, she did not accept the condition and decided to file a complaint against the company in the federal court of the US. Jami can also file complains for violating the privacy rights by the employer during the interview process in accordance with the Equal Employment Opportunity Commission. Hence, the Title VII of the Civil Rights Act of 1964 of the American law can be applied in the case of Jami Jensen. A PPLICABLE TORT CLAIMS In the United States, the federal tort claim provides compensation for the damages made to a property or any sort of injury arising from the activities perform by the federal officer or the employer (United States Court, â€Å"

Good, Bad and Disturbing News about the Future Essay Example for Free

Good, Bad and Disturbing News about the Future Essay It seems that the development of our technology has some price to pay. In his article entitled â€Å"Good, Bad and Disturbing News about the Future†, Dr. Malcolm Smith, a family life specialist addresses to parents the transformations that many teenagers are undergoing over the changing times. According to him, â€Å"Our young people are very connected, yet they feel very unattached† and with this loss of the sense of belongingness, the youth has resorted into various self-destructive behaviors such as violence and despair, among others. The youth may already have various means of connecting to other people but they do not have the attachment that people before used to have. Parents must be alarmed at the revelation that not many teenagers are able to communicate with adults well and this could lead to disobedience, which is, by the way, one of the Ten Commandments written in the Scriptures. The increasing disregard of the youth for others has caused them to rebel and go against what the teachings of their parents. In the end, this disregard leads them to lose their fear over the Supreme Being. While the whole world may be planning ways on how to survive future happenings such as recessions and the like, Malcolm said that it might also be high time to think over what can be done to change the youth’s attitude on life. A ministry can greatly help guide the youth such as through the retelling of the Scriptures, so that they have a chance to reflect on how they have lived their own life. This is a hard task at the beginning; however, as teenagers would only like to feel like they belong to some group, allowing them to interact with those who have similar situations can help them feel comfortable in no time. Bringing in family and ministry support to misguided teenagers could help prepare them for the future, helping them to survive and live life free from violence and rebellion.

Competitive Grants, Action Research Proposal, and Business Plans Essay Example for Free

Competitive Grants, Action Research Proposal, and Business Plans Essay Competitive grants, action research proposals, and business plans all seem to be predisposed with results and outcomes. All three seem to be geared towards the realization of a goal or an objective that supports desirable changes and results in society. Competitive grants are designed to be able to provide help and support to various populations who are in dire need of them (USDH, 2000). Action Research Proposals are written because of the need for changes in people’s behavior and responses toward certain issues or themes that need to be improved in order to result to desirable change through action research (Ferrance, 2000). Business plans are also designed in such a way that it provides for the needs of the consumers or the people and influence desirable changes in the economy and the satisfaction of the nation. It includes all the most important decision that one can make in order to make the business progress and be successful in its venture. (My Own Business, 2007)   Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚   In addition, competitive grants, action research proposals, and business plans are instrumental in identifying and determining the varied needs of members in the society. These three also differ in their purpose, content, and outline. To have a deeper understanding about the similarities and differences among these three, the discussion about their nature and characteristics is relevant and will be stated in the following paragraphs.   Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚   Funding programs for a specific group of population takes on the form of competitive grants. Competitive grants aim to address the growing needs of and problems in society. However, grants are only available to those who are eligible to apply and be granted access to the benefits and advantages that goes along with the grant. The funding and support that come from competitive grants are only limited to those who fit the characteristics of its target population. Moreover, these grants are only available for a limited period of time as proclaimed by the public or private organization implementing the program. Examples of competitive grants include training programs for young people who cannot afford to go to school, funding for nutrition and a healthy lifestyle advocacy, etc. (USDH, 2000)   Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚   An action research proposal plays a significant role in conducting a research, especially one that is academic in nature. The purpose of an action research proposal is to establish a background about the theme that is to be addressed after the research. In addition, an action research aims to influence change in certain processes and progressions that would in turn affect desirable changes in society. The action research proposal is instrumental in the accomplishment of this mission as it provides an overview of the problem and the solutions that would cause pleasant changes. The action research proposal is the basis of the action research. All the information and other action researches that have been conducted in the past are included in the proposal. In addition, it also contains predictions or calculated guesses of what the action research would lead to. It also includes the various processes that will be employed in order to collect all data, interpret them, and establish a solution to the main problem in the proposal. (Ferrance, 2000)   Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚   A business plan is a tool used by professionals in the business arena to predict the status of a specific business venture. It is also a way of preventing losses or cutbacks in the future. The business plan also continuously functions as a monitoring guide in evaluating or assessing the path of the business, whether it follows the plan or it is progressing or digressing in a different direction. Therefore, it is important to include substantial information and integrate it in the plan in order to see that changes that are occurring in a business enterprise. (Berry, 2008) The business plan is also broad than the action plan proposal and the competitive grant because it covers a wider range of information, from consumers, to human resources, capital funds, marketing strategies, etc. (My Own Business, 2007) References Berry, T. (2008). What is a Business Plan? Retrieved March 22, 2008, from Palo Alto Software, Inc. Website: http://articles.bplans.com/index.php/business-articles/writing-a-business-plan/What-is-a-Business-Plan?/ Ferrance, A. (2000). Action Research. Retrieved March 22, 2008, from The Education Alliance.   Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚   Website: http://www.alliance.brown.edu/pubs/themes_ed/act_research.pdf My Own Business. (2007). Business Plan. Retrieved March 22, 2008, from My Own Business,   Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚   Inc. Website: http://www.myownbusiness.org/s2/#1 USDH. (2000). What is a Competitive Grant Program. Retrieved March 22, 2008, from U.S.   Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚  Ã‚   Department of Housing. Website: http://www.hud.gov/nofa/suprnofa/sprprt1b.cfm

The real-time transport protocol

The real-time transport protocol Abstract This paper describes the Real-time Transport Protocol (RTP) with the emphasis on the securities, confidentiality and authenticity. This system takes a media file as input, encrypt it and create a message digest on the encrypted data then transmit it to the user. On the other side the receiver again calculate digest and compare it with the received one, if match occurs then decrypt and play it in the real time player. In current specification of RFC1889, only the confidentiality is described and authenticity is left for lower layer protocols. This work made experiment both on authenticity and confidentiality. For authenticity MD5, SHA-1 and SHA-2 hash algorithms can be used and for confidentiality AES-128 and Triple DES cryptographic algorithms can be used. In fact, SHA-2 is better than other hash algorithms in terms of security but SHA-1 is better than SHA-2 in terms of time efficiency. On the other hand, AES-128 is better than Triple DES in terms of time efficiency and security. So SHA-1and AES-128 is chosen for authenticity and confidentiality respectively for the security of RTP. The experiment is performed on J2SDK1.5. Keywords: Real-time transport protocol; Transport control protocol; Cryptographic algorithm; Hash algorithm. 1. Introduction: In recent days Computer and Internet has become essential part of human life and people demanding more and more access and use data over the Internet in Real-time with efficient secure manner. A new protocol Real-time Transport Protocol (RTP) and its associated Protocols helping people to use data over the Internet in their real-time applications. In this paper, the analyses of the securities of RTP, an approach to modify RTP for authenticity are presented. Also the position of RTP in Computer Network Layers, its usage scenario and the importance of time consideration to transmit streams using RTP are discussed and shown their analytical results. RTP is intended to be malleable to provide the information required by a particular application and will often be integrated into the application processing rather than being implemented as a separate layer. RTP is a modular protocol. The usage of RTP for a specific purpose requires an application area specific RTP profile. RTP profiles are used for refining the basic RTP protocol to suit for a particular application area. RTP profiles define how and by which formats data is encapsulated to RTP packets. RFC 1889 defines basic fields for the transportation of real time data. It also defines Real-time Transport Control Protocol (RTCP), whose purpose is to provide feedback on transmission quality, information about pa rticipants of RTP session, and enable minimal session control services. RTP is an application level protocol that is intended for delivery of delay sensitive content, such as audio and video, through different networks. The purpose of RTP is to facilitate delivery, monitoring, reconstruction, mixing and synchronization of data streams. RTP provides end-to-end network transport functions suitable for applications transmitting real-time data. RTP is a protocol framework that is deliberately not complete. RTP dose not provide quality of service means that it has no flow control, no error control, no acknowledgement and no mechanism to request retransmission. It does not do so because if a missing packet is retransmitted then it might happen that the retransmitted packet reach to the user too late to use which may hamper real-time use of streams. If some packets are lost during transmission (it is very common for Real-time protocols) then the lost packets are generated by interpolation rather than retransmission. However to improve performance of RTP another protocol; Real-time Transport Control Protocol is used with RTP. It handles feedback on delay, jitter, bandwidth, congestion, and other network properties. RTCP also handles inter stream synchronization. The problem is that different streams may use different clocks, with different granularities and different drift rates. RTCP can be used to keep them in synchronization. RTCP is also supports the use of RTP level translators and mixers. The paper is organized as follows. In section 2, RTP use scenarios are discussed. In section 3, position of RTP in computer network is discussed. In section 4, time consideration in RTP is discussed. In section 5, RTP packet format, its data transfer protocol and Real-time Transport control Protocol (RTCP) is discussed. Section 6 details the hash and cryptographic algorithms for RTP security while section 7 shows the result and performance analysis. Lastly, section 8 points out the conclusion inferred from the work. 2. RTP use scenarios: The following sections describe some aspects of the use of RTP. The examples are chosen to illustrate the basic operation of applications using RTP. In these examples, RTP is carried on top of IP and UDP and follows the conventions established by the profile for audio and video specified in the companion Internet-Draft draft-ietf-avt-profile. 2.1 Simple multicast audio conference: A working group of the IETF meets to discuss the latest protocol draft, using the IP multicast services of the Internet for voice communications. Through some allocation mechanism the working group chair obtains a multicast group address and pair of ports. One port is used for audio data, and the other is used for control (RTCP) packets. This address and port information is distributed to the intended participants. If privacy is desired, the data and control packets may be encrypted, in which case an encryption key must also be generated and distributed. The exact details of these allocation and distribution mechanisms are beyond the scope of RTP. The audio conferencing application used by each conference participant sends audio data in small chunks of, say, 20 ms duration. Each chunk of audio data is preceded by an RTP header; RTP header and data jointly form a UDP packet. The RTP header indicates what type of audio encoding (such as PCM, ADPCM or LPC) is contained in each packet so that senders can change the encoding during a conference, for example, to accommodate a new participant that is connected through a low-bandwidth link or react to indications of network congestion. The Internet, like other packet networks, occasionally loses and reorders packets and delays them by variable amounts of time. To cope with these impairments, the RTP header contains timing information and a sequence number that allow the receivers to reconstruct the timing produced by the source, so that in this example, chunks of audio are contiguously played out the speaker every 20 ms. This timing reconstruction is performed separately for each source of RTP packets in the conference. The sequence number can also be used by the receiver to estimate how many packets are being lost. Since members of the working group join and leave during the conference, it is useful to know who is participating at any moment and how well they are receiving the audio data. For that purpose, each instan ces of the audio application in the conference periodically multicasts a reception report plus the name of its user on the RTCP (control) port. The reception report indicates how well the current speaker is being received and may be used to control adaptive encoding. In addition to the user name, other identifying information may also be included subject to control bandwidth limits. A site sends the RTCP BYE packet when it leaves the conference. 2.2 Audio and Video Conference: If both audio and video media are used in a conference, they are transmitted as separate RTP sessions RTCP packets are transmitted for each medium using two different UDP port pairs and/or multicast addresses. There is no direct coupling at the RTP level between the audio and video sessions, except that a user participating in both sessions should use the same distinguished (canonical) name in the RTCP packets for both so that the sessions can be associated. One motivation for this separation is to allow some participants in the conference to receive only one medium if they choose. Despite the separation, synchronized playback of a sources audio and video can be achieved using timing information carried in RTCP packets for both sessions 2.3 Mixers and translators: So far, it is assumed that all sites want to receive media data in the same format. However, this may not always be appropriate. Consider the case where participants in one area are connected through a low-speed link to the majority of the conference participants who enjoy high-speed network access. Instead of forcing everyone to use a lower-bandwidth, reduced-quality audio encoding, an RTP-level relay called a mixer may be placed near the low-bandwidth area. This mixer resynchronizes incoming audio packets to reconstruct the constant 20 ms spacing generated by the sender, mixes these reconstructed audio streams into a single stream, translates the audio encoding to a lower-bandwidth one and forwards the lower-bandwidth packet stream across the low-speed link. These packets might be unicast to a single recipient or multicast on a different address to multiple recipients. The RTP header includes a means for mixers to identify the sources that contributed to a mixed packet so that corr ect talker indication can be provided at the receivers. Some of the intended participants in the audio conference may be connected with high bandwidth links but might not be directly reachable via IP multicast. For example, they might be behind an application-level firewall that will not let any IP packets pass. For these sites, mixing may not be necessary; in which case another type of RTP-level relay called a translator may be used. Two translators are installed, one on either side of the firewall, with the outside one funneling all multicast packets received through a secure connection to the translator inside the firewall. The translator inside the firewall sends them again as multicast packets to a multicast group restricted to the sites internal network. Mixers and translators may be designed for a variety of purposes. An example is a video mixer that scales the images of individual people in separate video streams and composites them into one video stream to simulate a group scene. Other examples of translation include the connection of a group of hosts speaking only IP/UDP to a group of hosts that understand only ST-II, or the packet-by-packet encoding translation of video streams from individual sources without resynchronization or mixing. 3. Position of RTP in computer network: As a consequence it is decided to put RTP in user space but should run over User Datagram Protocol (UDP, it is connectionless transport protocol). RTP operates as follows. The multimedia (streams) application consists of multiple audio, video, text and possibly other streams. These are fed into the RTP library, which is in user space along with the application. This library then multiplexes the streams and encodes those RTP packers, which it then stuffs into a socket. At the other end of socket (in the operating system kernel), UDP packets are generated and embedded in IP packets. If computer is on an Ethernet, the IP packets are then put in Ethernet frames for transmission. The protocol stack for this situation is shown in Figure 1. Since RTP runs in user space not in OS kernel and in the Ethernet it packed by UDP, then IP and then Ethernet, so it is difficult which layer RTP is in. But it is linked to application program and it is generic, application independent protocol that just provides transport facilities, so it is a transport protocol that is implemented in the application layer. The packet nesting is shown in Figure 2. The main goal of RTP is to make the transmitting streams real-time applicable. On the other hand if security is provided to the transmitting streams, then some extra time is needed to encrypt the streams or to make signature such as making digest from entire movie or audio files. So in concern of time, security is added with some overheads to the RTP. The goal of this paper is to choose algorithms and procedures that make RTP reliable in term of time and security. 4. Time consideration in RTP: Let a video or audio file over the Internet in real-time is intended to access, then here the most important parameter is bandwidth of the Network. And the next important parameters are minimum clip size and its duration as well as processors speed of both server and client. At first let it be assumed that files are accessing without security consideration. Then let review the following mathematical calculations for audio or video clip to access in real-time. One second file clip size = oneSecFileSize bits, Time duration of each clip = cSec seconds, Upload Transmission rate = uRate bits per second, Download Transmission rate = dRate bits per second, Time to upload, tUpload = oneSecFileSize *cSec/uRate, Time to download, tDownload = oneSecFileSize If the time to upload or download a clip is more than the time to play a clip, the player will wait and the receiver will see a break, i.e. max (tUpload, tDownload) >cSec. For the continuous playing of clips, the following condition must be true: Max (1/uRate, 1/dRate) > 1/ oneSecFileSize Min (uRate, dRate) > oneSecFileSize According to the equation, the waiting time between clips at the receiver does not depend on clip size. The only variable that matters for a continuous playback is the size of a one-second file and that the provided upload and download rates meet the above condition. Lag time between playing and capturing is: cSec + tupload + tdownload From the above equation, the maximum lag with no break in the feed is 3*cSec and the minimum lag is cSec. To get the clip as close to real time as possible, cSec should be reduced. Next, apply the above analysis to the following cases: 4.1 Both sender and receiver have a low bandwidth modem connection: Lets assume the uRate = dRate = 20K bits/sec. In this case, the one-second file size should be less than 20Kbits. If the clip size is 10 seconds, the maximum playback lag will be 30 seconds. It is observed that the minimum file size for transmitting a one-second video (with no audio) is 8Kbits using H263 encoding and 12896 pixels video size. It is also observed a minimum file size with the video and an 8-bit mono audio with an 8000Hz-sampling rate to be 80Kbits. 4.2 Either the sender or the receiver has a low bandwidth connection: Lets assume that the lower rate is 20Kbits/sec and the other rate is much higher. In this case the one-second file size should be less than 20Kbits, but the maximum playback lag is about 20 seconds if the clip size is 10 seconds. 4.3 Both sender and receiver have high bandwidth: It is noted here that the one-second-clip size may vary from the format to format of the file, that is, how the file is encoded. For example the one-second-clip size of MP3 is less than in WAV file. But the important point here is that when cryptographic algorithms are applied in the clip then an extra time is added to the processing of clip with each side. So if applied strong encryption algorithms to the clip then extra more time is needed to both sides and upload or download time will be affected and time lag between them will also be changed. So real time access of data is also affected. Therefore, providing security in RTP the considered parameters are bandwidth of the network, file format of clips, upload and download of the clip, processor and memory speed and applying cryptographic and hash algorithms. 5. RTP packet format and data transfer protocol: RTP packet formats and its Data Transfer Protocol is as follows: 5.1 RTP fixed header files: Whenever data are transferred with RTP, it always add a fixed header with the payload. The RTP header has the following format shown in figure 4: The first twelve octets are present in every RTP packet, while the list of CSRC identifiers is present only when inserted by a mixer. Version (V) is 2 bits wide. This field identifies the version of RTP. The version defined by this specification is two (2). Padding (P) is 1 bit wide. If the padding bit is set, the packet contains one or more additional padding octets at the end which are not part of the payload. The last octet of the padding contains a count of how many padding octets should be ignored. Padding may be needed by some encryption algorithms with fixed block sizes or for carrying several RTP packets in a lower-layer protocol data unit. Extension (X) is 1 bit wide. If the extension bit is set, the fixed header is followed by exactly one header extension. CSRC Count (CC) is 4 bits wide. The CSRC count contains the number of CSRC identifiers that follow the fixed header. Marker (M) is 1 bit wide. The interpretation of the marker is defined by a profile. It is intended to al low significant events such as frame boundaries to be marked in the packet stream. A profile may define additional marker bits or specify that there is no marker bit by changing the number of bits in the payload type field. Payload type (PT) is 7 bits wide. This field identifies the format of the RTP payload and determines its interpretation by the application. A profile specifies a default static mapping of payload type codes to payload formats. Additional payload type codes may be defined dynamically through non-RTP means. An initial set of default mappings for audio and video is specified in the companion profile Internet-Draft draft-ietf-avt-profile, and may be extended in future editions of the Assigned Numbers RFC [9]. An RTP sender emits a single RTP payload type at any given time; this field is not intended for multiplexing separate media streams. Sequence number is 16 bits wide. The sequence number increments by one for each RTP data packet sent, and may be used by the rece iver to detect packet loss and to restore packet sequence. The initial value of the sequence number is random (unpredictable) to make known-plaintext attacks on encryption more difficult, even if the source itself does not encrypt, because the packets may flow through a translator that does. Time stamp is 32 bits wide. The timestamp reflects the sampling instant of the first octet in the RTP data packet. The sampling instant must be derived from a clock that increments monotonically and linearly in time to allow synchronization and jitter calculations. The resolution of the clock must be sufficient for the desired synchronization accuracy and for measuring packet arrival jitter (one tick per video frame is typically not sufficient). The clock frequency is dependent on the format of data carried as payload and is specified statically in the profile or payload format specification that defines the format, or may be specified dynamically for payload formats defined through non-RTP mean s. If RTP packets are generated periodically, the nominal sampling instant as determined from the sampling clock is to be used, not a reading of the system clock. As an example, for fixed-rate audio the timestamp clock would likely increment by one for each sampling period. If an audio application reads the blocks covering 160 sampling periods from the input device, the timestamp would be increased by 160 for each such block, regardless of whether the block is transmitted in a packet or dropped as silent. The initial value of the timestamp is random, as for the sequence number. Several consecutive RTP packets may have equal timestamps if they are (logically) generated at once, e.g., belong to the same video frame. Consecutive RTP packets may contain timestamps that are not monotonic if the data is not transmitted in the order it was sampled, as in the case of MPEG interpolated video frames. SSRC is 32 bits wide. The SSRC field identifies the synchronization source. This identifier i s chosen randomly, with the intent that no two synchronization sources within the same RTP session will have the same SSRC identifier although the probability of multiple sources choosing the same identifier is low, all RTP implementations must be prepared to detect and resolve collisions. If a source changes its source transport address, it must also choose a new SSRC identifier to avoid being interpreted as a looped source. The CSRC list contains 0 to15 items, 32 bits each. The CSRC list identifies the contributing sources for the payload contained in this packet. The number of identifiers is given by the CC field. If there are more than 15 contributing sources, only 15 may be identified. CSRC identifiers are inserted by mixers, using the SSRC identifiers of contributing sources. For example, for audio packets the SSRC identifiers of all sources that were mixed together to create a packet is listed, allowing correct talker indication at the receiver. 5.2 Multiplexing RTP sessions: For efficient protocol processing, the number of multiplexing points should be minimized. In RTP, multiplexing is provided by the destination transport address (network address and port number), which defines an RTP session. For example, in a teleconference composed of audio and video media encoded separately, each medium should be carried in a separate RTP session with its own destination transport address. It is not intended that the audio and video be carried in a single RTP session and demultiplexed based on the payload type or SSRC fields. Interleaving packets with different payload types but using the same SSRC would introduce several problems: If one payload type were switched during a session, there would be no general means to identify which of the old values the new one replaced. An SSRC is defined to identify a single timing and sequence number space. Interleaving multiple payload types would require different timing spaces if the media clock rates differ and would require different sequence number spaces to tell which payload type suffered packet loss. The RTCP sender and receiver reports can only describe one timing and sequence number space per SSRC and do not carry a payload type field. An RTP mixer would not be able to combine interleaved streams of compatible media into one stream. Carrying multiple media in one RTP session precludes: the use of different network paths or network resource allocations if appropriate; reception of a subset of the media if desired, for example just audio if video would exceed the available bandwidth; and receiver implementations that use separate processes for the different media, whereas using separate RTP sessions permits either single- or multiple-process implementations. Using a different SSRC for each medium but sending them in the same RTP session would avoid the first three problems but not the last two. 5.3 Profile-Specific modifications to the RTP header: The existing RTP data packet header is believed to be complete for the set of functions required in common across all the application classes that RTP might support. However, in keeping with the ALF design principle, the header may be tailored through modifications or additions defined in a profile specification while still allowing profile-independent monitoring and recording tools to function. The marker bit and payload type field carry profile-specific information, but they are allocated in the fixed header since many applications are expected to need them and might otherwise have to add another 32-bit word just to hold them. The octet containing these fields may be redefined by a profile to suit different requirements, for example with a more or fewer marker bits. If there are any marker bits, one should be located in the most significant bit of the octet since profile-independent monitors may be able to observe a correlation between packet loss patterns and the marker bit. Addit ional information that is required for a particular payload format, such as a video encoding, should be carried in the payload section of the packet. This might be in a header that is always present at the start of the payload section, or might be indicated by a reserved value in the data pattern. If a particular class of applications needs additional functionality independent of payload format, the profile under which those applications operate should define additional fixed fields to follow immediately after the SSRC field of the existing fixed header. Those applications will be able to quickly and directly access the additional fields while profile-independent monitors or recorders can still process the RTP packets by interpreting only the first twelve octets. If it turns out that additional functionality is needed in common across all profiles, then a new version of RTP should be defined to make a permanent change to the fixed header. 5.4 RTP header extension: An extension mechanism is provided to allow individual implementation to experiment with new payload-format-independent functions that require additional information to be carried in the RTP data packet header. This mechanism is designed so that the header extension may be ignored by other interoperating implementations that have not been extended. Note that this header extension is intended only for limited use. Most potential uses of this mechanism would be done better another way, using the methods described in the previous section. For example, a profile-specific extension to the fixed header is less expensive to process because it is not conditional nor in a variable location. Additional information is required for a particular payload format should not use this header extension but should be carried in the payload section of the packet. If the X bit in the RTP header is one, a variable-length header extension is appended to the RTP header, following the CSRC list if present. Th e RTP header extension is shown in figure 5: The header extension contains a 16-bit length field that counts the number of 32-bit words in the extension, excluding the four-octet extension header (therefore zero is a valid length). Only a single extension may be appended to the RTP data header. To allow multiple interoperating implementations to each experiment independently with different header extensions, or to allow a particular implementation to experiment with more than one type of header extension, the first 16 bits of the header extension are left open for distinguishing identifiers or parameters. The format of these 16 bits is to be defined by the profile specification under which the implementations are operating. This RTP specification does not define any header extensions itself. 5.5 Real-time Transport Control Protocol (RTCP): RTCP is a coordination protocol of RTP. It provides some tasks to increase the performance of RTP. The RTP control protocol (RTCP) is based on the periodic transmission of control packets to all participants in the session, using the same distribution mechanism as the data packets. The underlying protocol must provide multiplexing of the data and control packets, for example using separate port numbers with UDP. RTCP performs four functions: The primary function is to provide feedback on the quality of the data distribution. This is an integral part of the RTPs role as a transport protocol and is related to the flow and congestion control functions of other transport protocols. The feedback may be directly useful for control of adaptive encodings, but experiments with IP multicasting have shown that it is also critical to get feedback from the receivers to diagnose faults in the distribution. Sending reception feedback reports to all participants allows one who is observing problems to evaluate whether those problems are local or global. With a distribution mechanism like IP multicast, it is also possible for an entity such as a network service provider who is not otherwise involved in the session to receive the feedback information and act as a third-party monitor to diagnose network problems. This feedback function is performed by the RTCP sender and receiver reports. RTCP carries a persistent transport-level identifier for an RTP source called the canonical name or CNAME. Since the SSRC identifier may change if a conflict is discovered or a program is restarted, receivers require the CNAME to keep track of each participant. Receivers also require the CNAME to associate multiple data streams from a given participant in a set of related RTP sessions, for example to synchronize audio and video. The first two functions require that all participants send RTCP packets; therefore the rate must be controlled in order for RTP to scale up to a large number of participants. By having each participant send its control packets to all the others, each can independently observe the number of participants. This number is used to calculate the rate at which the packets are sent. A fourth, optional function is to convey minimal session control information, for example participant identification to be displayed in the user interface. This is most likely to be useful in loosely controlled sessions where participants enter and leave without membership control or parameter negotiation. RTCP serves as a convenient channel to reach all the participants, but it is not necessarily expected to support all the control communication requirements of an application. A higher-level session control protocol, which is beyond the scope of this document, may be needed. Functions (i)-(iii) are mandatory when RTP is used in the IP multicast environment, and are recommended for all environments. RTP application designers are advised to avoid mechanisms that can only work in unicast mode and will not scale to larger numbers. 5.6 RTCP transmission interval: RTP is designed to allow an application to scale automatically over session sizes ranging from a few participants to thousands. For example, in an audio conference the data traffic is inherently self-limiting because only one or two people will speak at a time, so with multicast distribution the data rate on any given link remains relatively constant independent of the number of participants. However, the control traffic is not self-limiting. If the reception reports from each participant were sent at a constant rate, the control traffic would grow linearly with the number of participants. Therefore, the rate must be scaled down. For each session, it is assumed that the data traffic is subject to an aggregate limit called the session bandwidth to be divided among the participants. This bandwidth might be reserved and the limit enforced by the network, or it might just be a reasonable share. The session bandwidth may be chosen based or some cost or a priori knowledge of the available network bandwidth for the session. It is somewhat independent of the media encoding, but the encoding choice may be limited by the session bandwidth. The session bandwidth parameter is expected to be supplied by a session management application when it invokes a media application, but media applications may also set a default based on the single-sender data bandwidth for the encoding selected for the session. The application may also enforce bandwidth limits based on multicast scope rules or other criteria. Bandwidth calculations for control and data traffic include lower- layer transport and network protocols (e.g., UDP and IP) since that are what the resource reservation system would need to know. The application can also be expected to know which of these protocols are in use. Link level headers a

Explain Different Religious teachings on Life After Death :: essays research papers fc

Over two thousand years ago Saint Tiruvalluvar wrote that "Death is like falling asleep, and birth is like awakening from that sleep." The early Hindu’s did not believe in Heaven, nor did they ascribe to such desire. Their early teachings were that they would be reunited with Mother Nature. There was no yearning to live eternally – their prayers were for a healthy life. The notion of reincarnation and Heaven evolved over time. In the Hindu religion, righteousness and to be without sin is of paramount importance. They are the determining factors whether one goes to Heaven or Hell. The Hindu’s teach, â€Å"As it does and as it acts, so it becomes: The doer of good becomes good, and the doer of evil becomes evil†¦" Hindus moved onto believe in a cyclical time span. This means they believe people don not live and die just once but are reborn a number of times before reaching their final state. Hindus believe that after death the soul is reborn in this world to live a new life. What the person and into which caste they are reborn into is decided by there karma. Karma will try to increase their goof karma by keeping dharma which is their duty to the Gods. The process of being born growing, dying and being reborn is called samsara. It is the goal of every Hindu to be freed from this cycle in order to be in the presence of God or become on with God. The idea of reincarnation derives from the Bhagavad-Gita. There are three four verses imparticular, the first one talk about the soul not ever being born or dying that it is always alive and it is therefore the body which dies. â€Å"It (the self) is not born, and It does not die; nor is it ever that this One having been nonexistent becomes existent again. This One is birthless, eternal, undecaying, ancient; It is not killed when the body is killed.† - Gita Ch. 2 Verse 20 The second verse related to reincarnation states â€Å"As after rejecting (discarding) worn-out clothes a man takes up other new ones (clothes), likewise after rejecting worn-out bodies the embodied one (soul) duly attains new ones.† -Gita Ch.2 Verse 22. As if when the body is worn and grow as much as it can, the soul moves on. The third verse discusses that fact the reincarnation is a certainty.

Facts on Hamlet :: essays papers

Facts on Hamlet Title: Hamlet Published: It was written in the late 1500s. It has been on national stages since 1589, and the earliest known publications date back to the years 1600 and 1601. Biographical Information: Biographical Information: All though much is not known about Shakespeare the man, there is numerous speculations about Shakespeare the playwright. He was the son of John Shakespeare and Mary Arden. At the age of eighteen he married Anne Hathaway and had three children, Susanna, Hamlet, and Judith. William Shakespeare wrote thirty-seven plays and over one hundred and fifty poems which are renowned for their wisdom, knowledge of human nature, and genius Genre: Classical Drama Characteristics of a Genre: This is a dramatic tragedy, highlighting the fall of a great leader/character. Setting: The date in which the play took place is unknown, however it is believed to have been during the later parts of the 1500s during Shakespeare's life span. Although the exact date is unknown and to this day remains a mystery, the setting is revealed. The play underwent in Denmark and it's neighboring states. Characters: King Hamlet - The original king of Denmark who is Hamlet's father, and the husband of Gertrude. He is betrayed and murdered by his jealous brother, and later appears throughout the play to guide his son Hamlet in the quest for revenge and answers. He was Hamlet's father, and that was his relation to Hamlet. Hamlet - Son of Queen Gertrude, and King Hamlet, and later King Claudius (Who is his uncle). He is the main character who undergoes the most drastic changes of them all, psychologically. In the beginning he questioned all, but by the end after his journey had been cleared up he acted upon instinct and want, ultimately succeeding. Claudius - Jealous brother of King Hamlet, who acts upon his jealousy and dumps poison in the King (his brother's) ear and takes his life so that he may become king. He is now the father of Hamlet and the husband of Queen Gertrude. His jealousy and all around evil causes him to lose his life in the finality. He was Hamlet's uncle and new father, who was responsible for his real father's death.