7: Multimedia Networking7-1 Multimedia, Quality of Service: What is it? Multimedia applications: network audio and video (“continuous media”) network provides application with level of performance needed for application to function. QoS

7: Multimedia Networking7-2 Chapter 7: Goals Principles r Classify multimedia applications r Identify the network services the apps need r Making the best of best effort service r Mechanisms for providing QoS Protocols and Architectures r Specific protocols for best-effort r Architectures for QoS

7: Multimedia Networking7-3 Chapter 7 outline r 7.1 Multimedia Networking Applications r 7.2 Streaming stored audio and video r 7.3 Real-time Multimedia: Internet Phone study r 7.4 Protocols for Real- Time Interactive Applications m RTP,RTCP,SIP r 7.5 Distributing Multimedia: content distribution networks r 7.6 Beyond Best Effort r 7.7 Scheduling and Policing Mechanisms r 7.8 Integrated Services and Differentiated Services r 7.9 RSVP

7: Multimedia Networking7-4 MM Networking Applications Fundamental characteristics: r Typically delay sensitive m end-to-end delay m delay jitter r But loss tolerant: infrequent losses cause minor glitches r Antithesis of data, which are loss intolerant but delay tolerant. Classes of MM applications: 1) Streaming stored audio and video 2) Streaming live audio and video 3) Real-time interactive audio and video Jitter is the variability of packet delays within the same packet stream

7: Multimedia Networking7-5 Streaming Stored Multimedia Streaming: r media stored at source r transmitted to client r streaming: client playout begins before all data has arrived r timing constraint for still-to-be transmitted data: in time for playout

7: Multimedia Networking7-6 Streaming Stored Multimedia: What is it? 1. video recorded 2. video sent 3. video received, played out at client Cumulative data streaming: at this time, client playing out early part of video, while server still sending later part of video network delay time

7: Multimedia Networking7-7 Streaming Stored Multimedia: Interactivity r VCR-like functionality: client can pause, rewind, FF, push slider bar m 10 sec initial delay OK m 1-2 sec until command effect OK m RTSP often used (more later) r timing constraint for still-to-be transmitted data: in time for playout

7: Multimedia Networking7-8 Streaming Live Multimedia Examples: r Internet radio talk show r Live sporting event Streaming r playback buffer r playback can lag tens of seconds after transmission r still have timing constraint Interactivity r fast forward impossible r rewind, pause possible!

7: Multimedia Networking7-9 Interactive, Real-Time Multimedia r end-end delay requirements: m audio: < 150 msec good, < 400 msec OK includes application-level (packetization) and network delays higher delays noticeable, impair interactivity r session initialization m how does callee advertise its IP address, port number, encoding algorithms? r applications: IP telephony, video conference, distributed interactive worlds

7: Multimedia Networking7-10 Multimedia Over Today’s Internet TCP/UDP/IP: “best-effort service” r no guarantees on delay, loss Today’s Internet multimedia applications use application-level techniques to mitigate (as best possible) effects of delay, loss But you said multimedia apps requires QoS and level of performance to be effective! ? ? ?? ? ? ? ? ? ? ?

7: Multimedia Networking7-11 How should the Internet evolve to better support multimedia? Integrated services philosophy: r Fundamental changes in Internet so that apps can reserve end-to-end bandwidth r Requires new, complex software in hosts & routers Laissez-faire r no major changes r more bandwidth when needed r content distribution, application-layer multicast m application layer Differentiated services philosophy: r Fewer changes to Internet infrastructure, yet provide 1st and 2nd class service. What’s your opinion?

7: Multimedia Networking7-12 A few words about audio compression r Analog signal sampled at constant rate m telephone: 8,000 samples/sec m CD music: 44,100 samples/sec r Each sample quantized, i.e., rounded m e.g., 2 8 =256 possible quantized values r Each quantized value represented by bits m 8 bits for 256 values r Example: 8,000 samples/sec, 256 quantized values --> 64,000 bps r Receiver converts it back to analog signal: m some quality reduction Example rates r CD: Mbps r MP3: 96, 128, 160 kbps r Internet telephony: kbps

7: Multimedia Networking7-13 A few words about video compression r Video is sequence of images displayed at constant rate m e.g. 24 images/sec r Digital image is array of pixels r Each pixel represented by bits r Redundancy m spatial m temporal Examples: r MPEG 1 (CD-ROM) 1.5 Mbps r MPEG2 (DVD) 3-6 Mbps r MPEG4 (often used in Internet, < 1 Mbps) Research: r Layered (scalable) video m adapt layers to available bandwidth

7: Multimedia Networking7-14 Chapter 7 outline r 7.1 Multimedia Networking Applications r 7.2 Streaming stored audio and video r 7.3 Real-time Multimedia: Internet Phone study r 7.4 Protocols for Real- Time Interactive Applications m RTP,RTCP,SIP r 7.5 Distributing Multimedia: content distribution networks r 7.6 Beyond Best Effort r 7.7 Scheduling and Policing Mechanisms r 7.8 Integrated Services and Differentiated Services r 7.9 RSVP

7: Multimedia Networking7-15 Streaming Stored Multimedia Application-level streaming techniques for making the best out of best effort service: m client side buffering m use of UDP versus TCP m multiple encodings of multimedia r jitter removal r decompression r error concealment r graphical user interface w/ controls for interactivity Media Player

7: Multimedia Networking7-16 Internet multimedia: simplest approach audio, video not streamed: r no, “pipelining,” long delays until playout! r audio or video stored in file r files transferred as HTTP object m received in entirety at client m then passed to player

7: Multimedia Networking7-17 Internet multimedia: streaming approach r browser GETs metafile r browser launches player, passing metafile r player contacts server r server streams audio/video to player

7: Multimedia Networking7-18 Streaming from a streaming server r This architecture allows for non-HTTP protocol between server and media player r Can also use UDP instead of TCP.

7: Multimedia Networking7-19 constant bit rate video transmission Cumulative data time variable network delay client video reception constant bit rate video playout at client client playout delay buffered video Streaming Multimedia: Client Buffering r Client-side buffering, playout delay compensate for network-added delay, delay jitter

7: Multimedia Networking7-20 Streaming Multimedia: Client Buffering r Client-side buffering, playout delay compensate for network-added delay, delay jitter buffered video variable fill rate, x(t) constant drain rate, d

7: Multimedia Networking7-21 Streaming Multimedia: UDP or TCP? UDP r server sends at rate appropriate for client (oblivious to network congestion !) m often send rate = encoding rate = constant rate m then, fill rate = constant rate - packet loss r short playout delay (2-5 seconds) to compensate for network delay jitter r error recover: time permitting TCP r send at maximum possible rate under TCP r fill rate fluctuates due to TCP congestion control r larger playout delay: smooth TCP delivery rate r HTTP/TCP passes more easily through firewalls

7: Multimedia Networking7-22 Streaming Multimedia: client rate(s) Q: how to handle different client receive rate capabilities? m 28.8 Kbps dialup m 100Mbps Ethernet A: server stores, transmits multiple copies of video, encoded at different rates 1.5 Mbps encoding 28.8 Kbps encoding

7: Multimedia Networking7-23 Chapter 7 outline r 7.1 Multimedia Networking Applications r 7.2 Streaming stored audio and video r 7.3 Real-time Multimedia: Internet Phone case study r 7.4 Protocols for Real- Time Interactive Applications m RTP,RTCP,SIP r 7.5 Distributing Multimedia: content distribution networks r 7.6 Beyond Best Effort r 7.7 Scheduling and Policing Mechanisms r 7.8 Integrated Services and Differentiated Services r 7.9 RSVP

7: Multimedia Networking7-24 Interactive Multimedia: Internet Phone Introduce Internet Phone by way of an example r speaker’s audio: alternating talk spurts, silent periods. m 64 kbps during talk spurt r pkts generated only during talk spurts m 20 msec chunks at 8 Kbytes/sec: 160 bytes data r application-layer header added to each chunk. r Chunk+header encapsulated into UDP segment. r application sends UDP segment into socket every 20 msec during talkspurt.

7: Multimedia Networking7-25 Internet Phone: Packet Loss and Delay r network loss: IP datagram lost due to network congestion (router buffer overflow) r delay loss: IP datagram arrives too late for playout at receiver m delays: processing, queueing in network; end-system (sender, receiver) delays m typical maximum tolerable delay: 400 ms r loss tolerance: depending on voice encoding, losses concealed, packet loss rates between 1% and 10% can be tolerated.

7: Multimedia Networking7-26 constant bit rate transmission Cumulative data time variable network delay (jitter) client reception constant bit rate playout at client client playout delay buffered data Delay Jitter r Consider the end-to-end delays of two consecutive packets: difference can be more or less than 20 msec

7: Multimedia Networking7-27 Internet Phone: Fixed Playout Delay r Receiver attempts to playout each chunk exactly q msecs after chunk was generated. m chunk has time stamp t: play out chunk at t+q. m chunk arrives after t+q: data arrives too late for playout, data “lost” r Tradeoff for q: m large q: less packet loss m small q: better interactive experience

7: Multimedia Networking7-28 Fixed Playout Delay Sender generates packets every 20 msec during talk spurt. First packet received at time r First playout schedule: begins at p Second playout schedule: begins at p’

7: Multimedia Networking7-29 Adaptive Playout Delay, I Dynamic estimate of average delay at receiver: where u is a fixed constant (e.g., u =.01). r Goal: minimize playout delay, keeping late loss rate low r Approach: adaptive playout delay adjustment: m Estimate network delay, adjust playout delay at beginning of each talk spurt. m Silent periods compressed and elongated. m Chunks still played out every 20 msec during talk spurt.

7: Multimedia Networking7-30 Adaptive playout delay II Also useful to estimate the average deviation of the delay, v i : The estimates d i and v i are calculated for every received packet, although they are only used at the beginning of a talk spurt. For first packet in talk spurt, playout time is: where K is a positive constant. Remaining packets in talkspurt are played out periodically

7: Multimedia Networking7-31 Adaptive Playout, III Q: How does receiver determine whether packet is first in a talkspurt? r If no loss, receiver looks at successive timestamps. m difference of successive stamps > 20 msec -->talk spurt begins. r With loss possible, receiver must look at both time stamps and sequence numbers. m difference of successive stamps > 20 msec and sequence numbers without gaps --> talk spurt begins.

7: Multimedia Networking7-32 Chapter 7 outline r 7.1 Multimedia Networking Applications r 7.2 Streaming stored audio and video r 7.3 Real-time Multimedia: Internet Phone study r 7.4 Protocols for Real- Time Interactive Applications m RTP,RTCP,SIP r 7.5 Distributing Multimedia: content distribution networks r 7.6 Beyond Best Effort r 7.7 Scheduling and Policing Mechanisms r 7.8 Integrated Services and Differentiated Services r 7.9 RSVP

7: Multimedia Networking7-33 Real-Time Protocol (RTP) r RTP specifies a packet structure for packets carrying audio and video data r RFC r RTP packet provides m payload type identification m packet sequence numbering m timestamping r RTP runs in the end systems. r RTP packets are encapsulated in UDP segments r Interoperability: If two Internet phone applications run RTP, then they may be able to work together

7: Multimedia Networking7-34 RTP runs on top of UDP RTP libraries provide a transport-layer interface that extend UDP: port numbers, IP addresses payload type identification packet sequence numbering time-stamping

7: Multimedia Networking7-35 RTP Example r Consider sending 64 kbps PCM-encoded voice over RTP. r Application collects the encoded data in chunks, e.g., every 20 msec = 160 bytes in a chunk. r The audio chunk along with the RTP header form the RTP packet, which is encapsulated into a UDP segment. r RTP header indicates type of audio encoding in each packet m sender can change encoding during a conference. r RTP header also contains sequence numbers and timestamps.

7: Multimedia Networking7-36 RTP and QoS r RTP does not provide any mechanism to ensure timely delivery of data or provide other quality of service guarantees. r RTP encapsulation is only seen at the end systems: it is not seen by intermediate routers. m Routers providing best-effort service do not make any special effort to ensure that RTP packets arrive at the destination in a timely matter.

7: Multimedia Networking7-37 RTP Header Payload Type (7 bits): Indicates type of encoding currently being used. If sender changes encoding in middle of conference, sender informs the receiver through this payload type field. Payload type 0: PCM mu-law, 64 kbps Payload type 3, GSM, 13 kbps Payload type 7, LPC, 2.4 kbps Payload type 26, Motion JPEG Payload type 31. H.261 Payload type 33, MPEG2 video Sequence Number (16 bits): Increments by one for each RTP packet sent, and may be used to detect packet loss and to restore packet sequence.

7: Multimedia Networking7-38 RTP Header (2) r Timestamp field (32 bytes long). Reflects the sampling instant of the first byte in the RTP data packet. m For audio, timestamp clock typically increments by one for each sampling period (for example, each 125 usecs for a 8 KHz sampling clock) m if application generates chunks of 160 encoded samples, then timestamp increases by 160 for each RTP packet when source is active. Timestamp clock continues to increase at constant rate when source is inactive. r SSRC field (32 bits long). Identifies the source of the RTP stream. Each stream in a RTP session should have a distinct SSRC.

7: Multimedia Networking7-39 Real-Time Control Protocol (RTCP) r Works in conjunction with RTP. r Each participant in RTP session periodically transmits RTCP control packets to all other participants. r Each RTCP packet contains sender and/or receiver reports m report statistics useful to application r Statistics include number of packets sent, number of packets lost, interarrival jitter, etc. r Feedback can be used to control performance m Sender may modify its transmissions based on feedback

7: Multimedia Networking7-40 RTCP - Continued - For an RTP session there is typically a single multicast address; all RTP and RTCP packets belonging to the session use the multicast address. - RTP and RTCP packets are distinguished from each other through the use of distinct port numbers. - To limit traffic, each participant reduces his RTCP traffic as the number of conference participants increases.

7: Multimedia Networking7-41 RTCP Packets Receiver report packets: r fraction of packets lost, last sequence number, average interarrival jitter. Sender report packets: r SSRC of the RTP stream, the current time, the number of packets sent, and the number of bytes sent. Source description packets: r address of sender, sender's name, SSRC of associated RTP stream. r Provide mapping between the SSRC and the user/host name.

7: Multimedia Networking7-42 Synchronization of Streams r RTCP can synchronize different media streams within a RTP session. r Consider videoconferencing app for which each sender generates one RTP stream for video and one for audio. r Timestamps in RTP packets tied to the video and audio sampling clocks m not tied to the wall- clock time r Each RTCP sender-report packet contains (for the most recently generated packet in the associated RTP stream): m timestamp of the RTP packet m wall-clock time for when packet was created. r Receivers can use this association to synchronize the playout of audio and video.

7: Multimedia Networking7-43 RTCP Bandwidth Scaling r RTCP attempts to limit its traffic to 5% of the session bandwidth. Example r Suppose one sender, sending video at a rate of 2 Mbps. Then RTCP attempts to limit its traffic to 100 Kbps. r RTCP gives 75% of this rate to the receivers; remaining 25% to the sender r The 75 kbps is equally shared among receivers: m With R receivers, each receiver gets to send RTCP traffic at 75/R kbps. r Sender gets to send RTCP traffic at 25 kbps. r Participant determines RTCP packet transmission period by calculating avg RTCP packet size (across the entire session) and dividing by allocated rate.

7: Multimedia Networking7-44 Chapter 7 outline r 7.1 Multimedia Networking Applications r 7.2 Streaming stored audio and video r 7.3 Real-time Multimedia: Internet Phone study r 7.4 Protocols for Real- Time Interactive Applications m RTP,RTCP,SIP r 7.5 Distributing Multimedia: content distribution networks r 7.6 Beyond Best Effort r 7.7 Scheduling and Policing Mechanisms r 7.8 Integrated Services and Differentiated Services r 7.9 RSVP

7: Multimedia Networking7-45 Content distribution networks (CDNs) Content replication r Challenging to stream large files (e.g., video) from single origin server in real time r Solution: replicate content at hundreds of servers throughout Internet m content downloaded to CDN servers ahead of time m placing content “close” to user avoids impairments (loss, delay) of sending content over long paths m CDN server typically in edge/access network origin server in North America CDN distribution node CDN server in S. America CDN server in Europe CDN server in Asia

7: Multimedia Networking7-46 Content distribution networks (CDNs) Content replication r CDN (e.g., Akamai) customer is the content provider (e.g., CNN) r CDN replicates customers’ content in CDN servers. When provider updates content, CDN updates servers origin server in North America CDN distribution node CDN server in S. America CDN server in Europe CDN server in Asia

7: Multimedia Networking7-47 CDN example origin server ( r distributes HTML r replaces: with h ttp:// HTTP request for DNS query for HTTP request for Origin server CDNs authoritative DNS server Nearby CDN server CDN company (cdn.com) r distributes gif files r uses its authoritative DNS server to route redirect requests

7: Multimedia Networking7-48 Chapter 7 outline r 7.1 Multimedia Networking Applications r 7.2 Streaming stored audio and video r 7.3 Real-time Multimedia: Internet Phone study r 7.4 Protocols for Real- Time Interactive Applications m RTP,RTCP,SIP r 7.5 Distributing Multimedia: content distribution networks r 7.6 Beyond Best Effort r 7.7 Scheduling and Policing Mechanisms r 7.8 Integrated Services and Differentiated Services r 7.9 RSVP

7: Multimedia Networking7-49 Improving QOS in IP Networks Thus far: “making the best of best effort” Future: next generation Internet with QoS guarantees m RSVP: signaling for resource reservations m Differentiated Services: differential guarantees m Integrated Services: firm guarantees r simple model for sharing and congestion studies:

7: Multimedia Networking7-50 Principles for QOS Guarantees r Example: 1MbpsI P phone, FTP share 1.5 Mbps link. m bursts of FTP can congest router, cause audio loss m want to give priority to audio over FTP packet marking needed for router to distinguish between different classes; and new router policy to treat packets accordingly Principle 1

7: Multimedia Networking7-51 Principles for QOS Guarantees (more) r what if applications misbehave (audio sends higher than declared rate) m policing: force source adherence to bandwidth allocations r marking and policing at network edge: provide protection (isolation) for one class from others Principle 2

7: Multimedia Networking7-52 Principles for QOS Guarantees (more) r Allocating fixed (non-sharable) bandwidth to flow: inefficient use of bandwidth if flows doesn’t use its allocation While providing isolation, it is desirable to use resources as efficiently as possible Principle 3

7: Multimedia Networking7-53 Principles for QOS Guarantees (more) r Basic fact of life: can not support traffic demands beyond link capacity Call Admission: flow declares its needs, network may block call (e.g., busy signal) if it cannot meet needs Principle 4

7: Multimedia Networking7-54 Summary of QoS Principles Let’s next look at mechanisms for achieving this ….

7: Multimedia Networking7-55 Chapter 7 outline r 7.1 Multimedia Networking Applications r 7.2 Streaming stored audio and video r 7.3 Real-time Multimedia: Internet Phone study r 7.4 Protocols for Real- Time Interactive Applications m RTP,RTCP,SIP r 7.5 Distributing Multimedia: content distribution networks r 7.6 Beyond Best Effort r 7.7 Scheduling and Policing Mechanisms r 7.8 Integrated Services and Differentiated Services r 7.9 RSVP

7: Multimedia Networking7-56 Scheduling And Policing Mechanisms r scheduling: choose next packet to send on link r FIFO (first in first out) scheduling: send in order of arrival to queue m real-world example? m discard policy: if packet arrives to full queue: who to discard? Tail drop: drop arriving packet priority: drop/remove on priority basis random: drop/remove randomly

7: Multimedia Networking7-57 Scheduling Policies: more Priority scheduling: transmit highest priority queued packet r multiple classes, with different priorities m class may depend on marking or other header info, e.g. IP source/dest, port numbers, etc.. m Real world example?

7: Multimedia Networking7-58 Scheduling Policies: still more round robin scheduling: r multiple classes r cyclically scan class queues, serving one from each class (if available) r real world example?

7: Multimedia Networking7-59 Scheduling Policies: still more Weighted Fair Queuing: r generalized Round Robin r each class gets weighted amount of service in each cycle

7: Multimedia Networking7-60 Policing Mechanisms Goal: limit traffic to not exceed declared parameters Three common-used criteria: r (Long term) Average Rate: how many pkts can be sent per unit time (in the long run) m crucial question: what is the interval length: 100 packets per sec or 6000 packets per min have same average! r Peak Rate: e.g., 6000 pkts per min. (ppm) avg.; 1500 ppm peak rate r (Max.) Burst Size: max. number of pkts sent consecutively (with no intervening idle)

7: Multimedia Networking7-61 Policing Mechanisms Token Bucket: limit input to specified Burst Size and Average Rate. r bucket can hold b tokens r tokens generated at rate r token/sec unless bucket full r over interval of length t: number of packets admitted less than or equal to (r t + b).

7: Multimedia Networking7-62 Policing Mechanisms (more) r token bucket, WFQ combine to provide guaranteed upper bound on delay, i.e., QoS guarantee! WFQ token rate, r bucket size, b per-flow rate, R D = b/R max arriving traffic

7: Multimedia Networking7-63 Chapter 7 outline r 7.1 Multimedia Networking Applications r 7.2 Streaming stored audio and video r 7.3 Real-time Multimedia: Internet Phone study r 7.4 Protocols for Real- Time Interactive Applications m RTP,RTCP,SIP r 7.5 Distributing Multimedia: content distribution networks r 7.6 Beyond Best Effort r 7.7 Scheduling and Policing Mechanisms r 7.8 Integrated Services and Differentiated Services r 7.9 RSVP

7: Multimedia Networking7-64 IETF Integrated Services r architecture for providing QOS guarantees in IP networks for individual application sessions r resource reservation: routers maintain state info (a la VC) of allocated resources, QoS req’s r admit/deny new call setup requests: Question: can newly arriving flow be admitted with performance guarantees while not violated QoS guarantees made to already admitted flows?

7: Multimedia Networking7-65 Intserv: QoS guarantee scenario r Resource reservation m call setup, signaling (RSVP) m traffic, QoS declaration m per-element admission control m QoS-sensitive scheduling (e.g., WFQ) request/ reply

7: Multimedia Networking7-66 Call Admission Arriving session must : r declare its QOS requirement m R-spec: defines the QOS being requested r characterize traffic it will send into network m T-spec: defines traffic characteristics r signaling protocol: needed to carry R-spec and T- spec to routers (where reservation is required) m RSVP

7: Multimedia Networking7-67 Intserv QoS: Service models [rfc2211, rfc 2212] Guaranteed service: r worst case traffic arrival: leaky-bucket-policed source r simple (mathematically provable) bound on delay [Parekh 1992, Cruz 1988] Controlled load service: r "a quality of service closely approximating the QoS that same flow would receive from an unloaded network element." WFQ token rate, r bucket size, b per-flow rate, R D = b/R max arriving traffic

7: Multimedia Networking7-68 Chapter 7 outline r 7.1 Multimedia Networking Applications r 7.2 Streaming stored audio and video r 7.3 Real-time Multimedia: Internet Phone study r 7.4 Protocols for Real- Time Interactive Applications m RTP,RTCP,SIP r 7.5 Distributing Multimedia: content distribution networks r 7.6 Beyond Best Effort r 7.7 Scheduling and Policing Mechanisms r 7.8 Integrated Services and Differentiated Services r 7.9 RSVP

7: Multimedia Networking7-69 Signaling in the Internet connectionless (stateless) forwarding by IP routers best effort service no network signaling protocols in initial IP design + = r New requirement: reserve resources along end-to-end path (end system, routers) for QoS for multimedia applications r RSVP: Resource Reservation Protocol [RFC 2205] m “ … allow users to communicate requirements to network in robust and efficient way.” i.e., signaling ! r earlier Internet Signaling protocol: ST-II [RFC 1819]

7: Multimedia Networking7-70 RSVP Design Goals 1. accommodate heterogeneous receivers (different bandwidth along paths) 2. accommodate different applications with different resource requirements 3. make multicast a first class service, with adaptation to multicast group membership 4. leverage existing multicast/unicast routing, with adaptation to changes in underlying unicast, multicast routes 5. control protocol overhead to grow (at worst) linear in # receivers 6. modular design for heterogeneous underlying technologies

7: Multimedia Networking7-71 RSVP: does not… r specify how resources are to be reserved r rather: a mechanism for communicating needs r determine routes packets will take r that’s the job of routing protocols r signaling decoupled from routing r interact with forwarding of packets r separation of control (signaling) and data (forwarding) planes

7: Multimedia Networking7-72 RSVP: simple audio conference r H1, H2, H3, H4, H5 both senders and receivers r multicast group m1 r no filtering: packets from any sender forwarded r audio rate: b r only one multicast routing tree possible H2 H5 H3 H4 H1 R1 R2R3

7: Multimedia Networking7-73 Multimedia Networking: Summary r multimedia applications and requirements r making the best of today’s best effort service r scheduling and policing mechanisms r next generation Internet: Intserv, RSVP, Diffserv