Multimedia Networking - PDF Free Download

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

Goals Principles Classify multimedia applications Identify the network services the apps need Making the best of best effort service Mechanisms for providing QoS Protocols and Architectures Specific protocols for best-effort Architectures for QoS VoIP services Specific architectures and protocols Implementation issues

MM Networking Applications 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 Fundamental characteristics: Typically delay sensitive end-to-end delay delay jitter But loss tolerant: infrequent losses cause minor glitches Antithesis of data, which are loss intolerant but delay tolerant.

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

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

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

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

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

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

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

A few words about audio compression Analog signal sampled at constant rate telephone: 8,000 samples/sec CD music: 44,100 samples/sec Each sample quantized, i.e., rounded e.g., 2 8 =256 possible quantized values Each quantized value represented by bits 8 bits for 256 values Example: 8,000 samples/sec, 256 quantized values --> 64,000 bps Receiver converts it back to analog signal: some quality reduction Example rates CD: 1.411 Mbps MP3: 96, 128, 160 kbps Internet telephony: 5.3-13 kbps

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

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

Internet multimedia: simplest approach audio or video stored in file files transferred as HTTP object received in entirety at client then passed to player audio, video not streamed: no, pipelining, long delays until playout!

Internet multimedia: streaming approach browser GETs metafile browser launches player, passing metafile player contacts server server streams audio/video to player

Streaming from a streaming server This architecture allows for non-http protocol between server and media player Can also use UDP instead of TCP.

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

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

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

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

User Control of Streaming Media: RTSP HTTP Does not target multimedia content No commands for fast forward, etc. RTSP: RFC 2326 Client-server application layer protocol. For user to control display: rewind, fast forward, pause, resume, repositioning, etc What it doesn t do: does not define how audio/video is encapsulated for streaming over network does not restrict how streamed media is transported; it can be transported over UDP or TCP does not specify how the media player buffers audio/video

RTSP: out of band control FTP uses an out-of-band control channel: A file is transferred over one TCP connection. Control information (directory changes, file deletion, file renaming, etc.) is sent over a separate TCP connection. The out-of-band and inband channels use different port numbers. RTSP messages are also sent out-of-band: RTSP control messages use different port numbers than the media stream: out-of-band. Port 554 The media stream is considered in-band.

RTSP Example Scenario: metafile communicated to web browser browser launches player player sets up an RTSP control connection, data connection to streaming server

Metafile Example <title>twister</title> <session> </session> <group language=en lipsync> <switch> <track type=audio e="pcmu/8000/1" src = "rtsp://audio.example.com/twister/audio.en/lofi"> <track type=audio e="dvi4/16000/2" pt="90 DVI4/8000/1" src="rtsp://audio.example.com/twister/audio.en/hifi"> </switch> <track type="video/jpeg" </group> src="rtsp://video.example.com/twister/video">

RTSP Operation

RTSP Exchange Example C: SETUP rtsp://audio.example.com/twister/audio RTSP/1.0 Cseq=1; Transport: rtp/udp; compression; port=3056; mode=play S: RTSP/1.0 200 OK Cseq=1 Session 4231 C: PLAY rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 Cseq=2; Session: 4231 Range: npt=0- S: RTSP/1.0 200 OK Cseq=2 Session 4231 C: PAUSE rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 Cseq=3; Session: 4231 Range: npt=37 S: RTSP/1.0 200 OK Cseq=3; Session 4231 C: TEARDOWN rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 Cseq=4; Session: 4231 S: RTSP/1.0 200 OK Cseq=4; Session 4231

Real-time interactive applications PC-2-PC phone instant messaging services are providing this PC-2-phone Dialpad Net2phone videoconference with Webcams Going to now look at a PC-2-PC Internet phone example in detail

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

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

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

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

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 packets packets generated packets received loss playout schedule p' - r playout schedule p - r time r p p'

Adaptive Playout Delay, I Goal: minimize playout delay, keeping late loss rate low Approach: adaptive playout delay adjustment: Estimate network delay, adjust playout delay at beginning of each talk spurt. Silent periods compressed and elongated. Chunks still played out every 20 msec during talk spurt. t r p r i i i d timestamp of the ith packet the time packet i is received by receiver i the time packet i is playedat receiver t i i network delay for ith packet estimate of average network delay after receiving ith packet Dynamic estimate of average delay at receiver: d i ( 1 u) di 1 u( ri ti ) where u is a fixed constant (e.g., u =.01).

Adaptive playout delay II Also useful to estimate the average deviation of the delay, v i : v i ( 1 u) vi 1 u ri ti di 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: i i d where K is a positive constant. p t Remaining packets in talkspurt are played out periodically i Kv i

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

Recovery from packet loss (1) forward error correction (FEC): simple scheme for every group of n chunks create a redundant chunk by exclusive OR-ing the n original chunks send out n+1 chunks, increasing the bandwidth by factor 1/n. can reconstruct the original n chunks if there is at most one lost chunk from the n+1 chunks Playout delay needs to be fixed to the time to receive all n+1 packets Tradeoff: increase n, less bandwidth waste increase n, longer playout delay increase n, higher probability that 2 or more chunks will be lost

Recovery from packet loss (2) 2nd FEC scheme piggyback lower quality stream send lower resolution audio stream as the redundant information for example, nominal stream PCM at 64 kbps and redundant stream GSM at 13 kbps. Whenever there is non-consecutive loss, the receiver can conceal the loss. Can also append (n-1)st and (n-2)nd low-bit rate chunk

Recovery from packet loss (3) Interleaving chunks are broken up into smaller units for example, 4 5 msec units per chunk Packet contains small units from different chunks if packet is lost, still have most of every chunk has no redundancy overhead but adds to playout delay

Real-Time Protocol (RTP) RTP specifies a packet structure for packets carrying audio and video data RFC 1889. RTP packet provides payload type identification packet sequence numbering timestamping RTP runs in the end systems. RTP packets are encapsulated in UDP segments Interoperability: If two Internet phone applications run RTP, then they may be able to work together

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

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

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

RTP header 0 2 3 4 8 9 16 24 31 V P X CC M PT Sequence Number Timestamp SSRC CSRC_1

RTP header version (V) = (2 bit) RTP protocol version; Padding (P) = (1 bit) Indicates the presence of padding bytes beyond data bytes; Extension (X) = (1 bit) Indicates the presence of an extension header; CSRC count (CC) = (4 bit) Number of CSRC fields in the header, i.e. number of sources that have generated data included in paylaod; Marker (M) = (1 bit) May be used by applications, for example to indicate end of data.

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. At the session start its value is set randomly, so that the probability of mixing packets of different sessions is minimized.

RTP header Timestamp field (32 bytes long). Reflects the sampling instant of the first byte in the RTP data packet. For audio, timestamp clock typically increments by one for each sampling period (for example, each 125 usecs for a 8 KHz sampling clock) 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. SSRC field (32 bits long). Identifies the source of the RTP stream. In case of multiple source, the SSRC source has mixed data. Contributing source (CSRC) = up to 15 fields, 32 bits each. - optional fields; - they include the SSRC of the true flow sources.

RTP header 32 bit Source port # Destination port # V P X C C Lenght Checksum (opt.) M PType Sequence number Timestamp Synchronization source (SSRC) identifier Possible header extension UDP header 8 B RTP header 12 B Payload

RTP header 0 8 16 24 31 Defined by profile Lenght header extension.. header extension used for individual implementations to test new features, payload formats, requiring information that cannot be included in the normal RTP header. lenght : header extension length expressed in 4 bytes words.

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

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.

RTCP Packets Receiver report packets Sender report packets Source description packets: e-mail address of sender, sender's name, SSRC of associated RTP stream. Provide mapping between the SSRC and the user/host name.

RTCP Packets SR (Sender Report): - sent from all active sources to all participants, i.e. SSRC of the RTP stream; - it includes transmission statistics collected by SSRCs - it includes information related to data sent: a) timestamp (NTP) of the sending time; b) timestamp relevant to the ongoing RTP flow; c) data sent since the session start: - total number of RTP packets sent; - total number of bytes sent.

RTCP Packets RR (Receiver Report): - sent from all passive terminals to all participants; - it includes reception statistics collected by a partecipanti receiving RTP data; - used to inform senders of the reception quality; - sent to all sources form which a SR has been received; - it includes: a) Indication of the transmistting suorce; b) Timestamp of the latter SR received; c) Reception delay of the latter SR; d) Highest sequence number received from the source; e) Number of lost RTP packets in the session; f) Fraction of lost RTP packets in the session; g) jitter estimate of RTP packets in the session.

RTCP Packets SDES (Source Descriptor): - description of RTP participants, - provide mapping between the SSRC and the user/host name; - unique identifier; - used by sources and receivers to present themselves. - it may include: a) CNAME: user id(user@host.domain); b) NAME: name of the person using the appliation; c) EMAIL; d) PHONE; e) LOC: user geographical location; f) TOOL: application using RTP; g) NOTE. BYE: - it can indicate a participant disconnection or the session end. APP: application-specific - it indicates that a participant wants to leave the session.

RTCP Packets Encription prefix 32 bit SR o RR Additional RRs SDES (CNAME) APP BYE

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

Problem!!! RTCP Bandwidth Scaling Consider an RTP session, with one sender and many receivers; Each receiver generates RTCP packets; the aggregate receiver data transmission rate may be higher than the sender data transmission rate. The amount of RTP traffic sent through the multicast tree does not change with the receiver number; The amount of RTCP traffic increases linearly with the receiver number. Solution: RTCP adapts the transmission rate of the session participants.

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

RTCP Bandwidth Scaling Each participant (sender or receiver) determines the transmission time of an RTCP packet by evaluation dinamically the average RTCP packet size and dividing it by the allocated transmissio rate. Ts = number of senders (ave. RTCP packet size) 0.25 x 0.05 x session bandwidth Tr = number of receivers (ave. RTCP packet size) 0.75 x 0.05 x session bandwidth

SIP

SIP Session Initiation Protocol Comes from IETF SIP long-term vision All telephone calls and video conference calls take place over the Internet People are identified by names or e-mail addresses, rather than by phone numbers. You can reach the callee, no matter where the callee roams, no matter what IP device the callee is currently using.

SIP Services Setting up a call Provides mechanisms for caller to let callee know she wants to establish a call Provides mechanisms so that caller and callee can agree on media type and encoding. Provides mechanisms to end call. Determine current IP address of callee. Maps mnemonic identifier to current IP address Call management Add new media streams during call Change encoding during call Invite others Transfer and hold calls

SIP: IETF standard H.323 vs SIP Derived from HTTP style signaling, Simple and interfaces well with IP networks, instant messaging (IM) Services are not explicitly exposed to protocol Well-defined methods can be used to design services: most telephony services have analogs in the SIP world today SIP is gathering market share rapidly

SIP Audio Codec G.711 G.723 G.729 Video Codec H.261 H.263 SIP RTP RTCP TCP UDP IP LAN Interface

SIP functionality IETF-standardized peer-to-peer signaling protocol (RFC 2543): Locate user given email-style address Setup session (call) (Re)-negotiate call parameters Manual and automatic forwarding Personal mobility: different terminal, same identifier Terminate and transfer calls

IP SIP Phones and Adaptors Are true Internet hosts Choice of application Choice of server IP appliances Implementations 3Com (3) Columbia University MIC WorldCom (1) Mediatrix (1) Nortel (4) Siemens (5) 4 2 3 1 Analog phone adaptor 5 Palm control

SIP components UAC: user-agent client (caller application) UAS: user-agent server: accept, redirect, refuse call redirect server: redirect requests proxy server: server + client registrar: track user locations user agent = UAC + UAS often combine registrar + (proxy or redirect server)

Setting up a call to a known IP address Alice 167.180.112.24 INVITE bob@193.64.210.89 c=in IP4 167.180.112.24 m=audio 38060 RTP/AVP 0 port 5060 port 5060 200 OK c=in IP4 193.64.210.89 m=audio 48753 RTP/AVP 3 ACK port 5060 193.64.210.89 Bob Bob's terminal rings Alice s SIP invite message indicates her port number & IP address. Indicates encoding that Alice prefers to receive (PCM ulaw) Bob s 200 OK message indicates his port number, IP address & preferred encoding (GSM) m Law audio port 38060 GSM port 48753 SIP messages can be sent over TCP or UDP; here sent over RTP/UDP. time time Default SIP port number is 5060.

Setting up a call (more) Codec negotiation: Suppose Bob doesn t have PCM ulaw encoder. Bob will instead reply with 606 Not Acceptable Reply and list encoders he can use. Alice can then send a new INVITE message, advertising an appropriate encoder. Rejecting the call Bob can reject with replies busy, gone, payment required, forbidden. Media can be sent over RTP or some other protocol.

Example of SIP message INVITE sip:bob@domain.com SIP/2.0 Via: SIP/2.0/UDP 167.180.112.24 From: sip:alice@hereway.com To: sip:bob@domain.com Call-ID: a2e3a@pigeon.hereway.com Content-Type: application/sdp Content-Length: 885 c=in IP4 167.180.112.24 m=audio 38060 RTP/AVP 0 Notes: HTTP message syntax sdp = session description protocol Call-ID is unique for every call. Here we don t know Bob s IP address. Intermediate SIP servers will be necessary. Alice sends and receives SIP messages using the SIP default port number 506. Alice specifies in Via: header that SIP client sends and receives SIP messages over UDP

Name translation and user locataion Caller wants to call callee, but only has callee s name or e-mail address. Need to get IP address of callee s current host: user moves around DHCP protocol user has different IP devices (PC, PDA, car device) Result can be based on: time of day (work, home) caller (don t want boss to call you at home) status of callee (calls sent to voicemail when callee is already talking to someone) Service provided by SIP servers: SIP registrar server SIP proxy server

SIP Registrar When Bob starts SIP client, client sends SIP REGISTER message to Bob s registrar server (similar function needed by Instant Messaging) Register Message: REGISTER sip:domain.com SIP/2.0 Via: SIP/2.0/UDP 193.64.210.89 From: sip:bob@domain.com To: sip:bob@domain.com Expires: 3600

SIP: Personal Mobility Users maintain a single externally visible identifier regardless of their network location

SIP Proxy Alice sends invite message to her proxy server contains address sip:bob@domain.com Proxy responsible for routing SIP messages to callee possibly through multiple proxies. Callee sends response back through the same set of proxies. Proxy returns SIP response message to Alice contains Bob s IP address Note: proxy is analogous to local DNS server

Example Caller jim@umass.edu with places a call to keith@upenn.edu (1) Jim sends INVITE message to umass SIP proxy. (2) Proxy forwards request to upenn registrar server. (3) upenn server returns redirect response, indicating that it should try keith@eurecom.fr SIP proxy umass.edu 1 8 SIP client 217.123.56.89 2 3 SIP registrar upenn.edu 4 7 9 SIP registrar eurecom.fr 6 5 SIP client 197.87.54.21 (4) umass proxy sends INVITE to eurecom registrar. (5) eurecom registrar forwards INVITE to 197.87.54.21, which is running keith s SIP client. (6-8) SIP response sent back (9) media sent directly between clients. Note: also a SIP ack message, which is not shown.

SIP as Event Notification Protocol

SIP for instant messaging: IM (RFC 3428) IM: transfer of (short) messages in near real-time, for conversational mode. Current IM: proprietary, server-based and linked to buddy lists etc MESSAGE method: inherits SIP s request routing and security features Message content as MIME body parts Sent in the context of some SIP dialog (note: slightly different from pager mode: asynchronous) Sent over TCP (or congestion controlled transports): lots of messaging volumes Allows IM applications to potentially interoperate and also provides SIP-based integration with other multimedia streams.

SIP: Presence

SIP-based Architecture Telephone Telephone switch sipconf SIP conference server rtspd RTSP media server sipum SIP/RTSP Unified messaging RTSP Quicktime RTSP clients Web based configuration T1/E1 RTP/SIP sipd SIP proxy, redirect server SQL database e*phone Web server Cisco 2600 gateway sipc Hardware Internet (SIP) phones sip323 NetMeeting Software SIP user agents SIPH.323 convertor H.323

Example Call Bob signs up for the service from the web as bob@ecse.rpi.edu He registers from multiple phones Alice tries to reach Bob INVITE ip:bob.wilson@ecse.rpi.edu Call Bob sipd SIP proxy, redirect server sipd canonicalizes the destination to sip:bob@ecse.rpi.edu sipd rings both e*phone and sipc Bob accepts the call from sipc and starts talking SQL database Web server e*phone Web based configuration sipc Hardware Internet (SIP) phones ecse.rpi.edu Software SIP user agents

george.w.bush parliament.uk Proxy Server 1. INVITE sip:president@us.gov SIP/2.0 From: sip:tony@parliament.uk 2. INVITE sip:dcheney@wh SIP/2.0 From: sip:tony@parliament.uk 3. SIP/2.0 200 ok From: sip:dcheney@wh 1 & 5 Location Server us.gov tony@parliament.uk 4 2 & 6 dcheney@wh 4. SIP/2.0 100 OK From: sip:president@us.gov 5. ACK sip:president@us.gov SIP/2.0 From: sip:tony@parliament.uk 6. ACK sip:dcheney@wh SIP/2.0 From: sip:tony@parliament.uk Proxy server 3

george.w.bush Redirect Server us.gov parliament.uk 1 & 3 Location Server tony@parliament.uk 2 dcheney@wh.us.gov 4 & 6 Redirect Server 1. INVITE sip:president@us.gov From: sip:tony@parliament.uk 2. SIP/2.0 320 Moved temporarily Contact: sip:dcheney@wh.us.gov 3. ACK sip:president@us.gov From: sip:tony@parliament.uk 5 4. INVITE sip:dcheney@wh.us.gov 6. ACK sip:dcheney@wh.us.gov From: tony@parliament.uk From: sip:tony@parliament.uk 5. SIP/2.0 200 OK To: tony@parliament.uk

SIP Call Signaling Assumes Endpoints(Clients) know each other s IP addresses SIP Endpoint SIP Gateway Signaling Plane Invite 180 Ringing 200 OK Ack SIP + SDP (TCP or UDP) Bearer Plane RTP Stream RTP Stream RTCP Stream Media (UDP)

SDP: Session Description Protocol RFC 2327 Not really a protocol describes data carried by other protocols Used by SAP, SIP, RTSP, H.332, PINT. Session description v= (protocol version) o= (owner/creator and session identifier). s= (session name) i=* (session information) u=* (URI of description) e=* (email address) p=* (phone number) c=* (connection information - not required if included in all media) b=* (bandwidth information) One or more time descriptions z=* (time zone adjustments) k=* (encryption key) a=* (zero or more session attribute lines) Zero or more media descriptions) Time description t= (time the session is active) r=* (zero or more repeat times) Media description m= (media name and transport address) i=* (media title) c=* (connection information - optional if included at session-level) b=* (bandwidth information) k=* (encryption key) a=* (zero or more media attribute lines)

SDP: Session Description Protocol RFC 2327 Example: o=<username> <session id> <version> < network type > <address type> <address> version number f or this announcement v=0 o=mhandley 2890844526 2890842807 IN IP4 126.16.64.4 s=sdp Seminar i=a Seminar on the session description protocol u=http://www.cs.ucl.ac.uk/staff/m.handley/sdp.03.ps e=mjh@isi.edu (Mark Handley) c=in IP4 224.2.17.12/127 t=2873397496 2873404696 a=recvonly m=audio 49170 RTP/AVP 0 m=video 51372 RTP/AVP 31 m=application 32416 udp wb a=orient:portrait internet URI should be a pointer to additional information about the conference Start and End time in NTP format

SIP Dialogs (RFC 3261) A dialog represents a peer-to-peer SIP relationship between two user agents that persists for some time. The dialog facilitates sequencing of messages between the user agents and proper routing of requests between both of them. The dialog represents a context in which to interpret SIP messages. A dialog is identified at each UA with a dialog ID, which consists of a Call-ID value, a local tag and a remote tag. A dialog contains certain pieces of state needed for further message transmissions within the dialog.

UPDATE method (RFC 3311) INVITE method: initiation and modification of sessions. INVITE affects two pieces of state: session (the media streams SIP sets up) and dialog (the state that SIP itself defines). Issue: need to modify session aspects before the initial INVITE has been answered. A re-invite cannot be used for this purpose: impacts the state of the dialog, in addition to the session. Ans: The UPDATE method Operation: (Offer/Answer model) The caller begins with an INVITE transaction, which proceeds normally. Once a dialog is established, either early or confirmed, the caller can generate an UPDATE method that contains an SDP offer for the purposes of updating the session. The response to the UPDATE method contains the answer. Similarly, once a dialog is established, the callee can send an UPDATE offer

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

CDN example HTTP request for www.foo.com/sports/sports.html 1 2 3 Origin server DNS query for www.cdn.com CDNs authoritative DNS server HTTP request for www.cdn.com/www.foo.com/sports/ruth.gif Nearby CDN server origin server (www.foo.com) distributes HTML replaces: http://www.foo.com/sports.ruth.gif with http://www.cdn.com/www.foo.com/sports/ruth.gif CDN company (cdn.com) distributes gif files uses its authoritative DNS server to route redirect requests

More about CDNs routing requests CDN creates a map, indicating distances from leaf ISPs and CDN nodes when query arrives at authoritative DNS server: server determines ISP from which query originates uses map to determine best CDN server CDN nodes create application-layer overlay network

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

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

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

Principles for QOS Guarantees (more) Allocating fixed (non-sharable) bandwidth to flow: inefficient use of bandwidth if flows doesn t use its allocation Principle 3 While providing isolation, it is desirable to use resources as efficiently as possible

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

Summary of QoS Principles Let s next look at mechanisms for achieving this.

Scheduling And Policing Mechanisms scheduling: choose next packet to send on link FIFO (first in first out) scheduling: send in order of arrival to queue real-world example? 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

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

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

Scheduling Policies: still more Weighted Fair Queuing: generalized Round Robin each class gets weighted amount of service in each cycle real-world example?

Policing Mechanisms Goal: limit traffic to not exceed declared parameters Three common-used criteria: (Long term) Average Rate: how many pkts can be sent per unit time (in the long run) crucial question: what is the interval length: 100 packets per sec or 6000 packets per min have same average! Peak Rate: to be defined according a reference time slot. (Max.) Burst Size: max. number of pkts sent consecutively (at peak rate)

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

Provisioning and Monitoring LB LB B TS P s B TS r s r s P s

Dual Leaky-Bucket b c B TS r s P s b P P s s c r s B TS b B TS r s P s c

Multiplexing DLB 1 B DLB 2 C... DLB k b Ex. Fair-Share C c B b c

Equivalent Bandwidth and Buffer D max c b b B c C or b B r P c P TS s s s B TS b c P s r s c 0 b 0 0 0 max max 0 max 0 max 0 ) ( b B c C n C B D c D b B r P D P B c b B c C TS S S TS S max 0 0 max 0 max 0 max 0 max, min ) ( D C B typically c C b B n c D b B r P D P B c D c b TS S S TS S Fair-Share Max delay

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

IETF Integrated Services architecture for providing QOS guarantees in IP networks for individual application sessions resource reservation: routers maintain state info (a la VC) of allocated resources, QoS req s 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?

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

Integrated Services - Principles Flow specification Tell the network what the flow wants e.g. 100 msec guaranteed to www.nsf.gov Admission control Network decides if it can handle flow Spec travels down path for approval Delay guarantee approved by all routers, so admitted Reservation Enable admission control Packet classification Map packets to flows e.g. packets marked as guaranteed Scheduling Forwarding policy e.g. guaranteed packets sent first

Intserv QoS: Service models [RFC 2211, RFC 2212] Guaranteed service: worst case traffic arrival: leaky-bucket-policed source simple (mathematically provable) bound on delay [Parekh 1992, Cruz 1988] Controlled load service: "a quality of service closely approximating the QoS that same flow would receive from an unloaded network element." arriving traffic token rate, r bucket size, b WFQ per-flow rate, R D = b/r max

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

Advanced IP Architecture RSVP Real Time Applications RTP/RTCP UDP IPv4/IPv6 Elastic Applications TCP Underlying Data Link Technologies

Role of RSVP Rides on top of unicast/multicast routing protocols Must be present at sender(s), receiver(s), and routers Carries resource requests all the way through the network At each hop consults admission control and sets up reservation

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

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

Flow Specification Session must first declare its QoS requirement and characterize the traffic it will send through the network R-spec defines the QoS being requested by receiver: Min Path Latency, Min Bdw, break bit, Hops, min MTU. T-spec defines the traffic characteristics of sender: Ps, rs, max burst size, min policed unit, max packt size. RSVP is the signaling protocol is needed to carry the R-spec and T-spec to the routers

Filter Specification The router needs to recognize the packets belonging to that flow IP of the sender IP destination Port number generating the packets Port number of the receiver Protocol ID Any field of the header flowspec + filterspec = flowdescriptor

PATH Messages PATH messages carry sender s Traffic Specifications (TSpec) Carries also the FilterSpec Routers take note of the PATH sender and set up reverse path to it Receivers send RESV messages that follow reverse path and setup reservations If reservation cannot be made, user gets an error

RESV Messages RESV messages carry receiver s QoS needs (Rspec) Forwarded via reverse path of PATH Queuing delay and bandwidth requirements Source traffic characteristics (from PATH) Filter specification Which transmissions can use the reserved resources? Reservation style. Router performs admission control and reserves resources

Router Handling of RESV Messages If new request rejected, send error message. If admitted: Install packet filter into forwarding dbase. Pass flow parameters to scheduler. Activate packet policing if needed. Forward RESV message upstream.

RSVP Functional Diagram Host Router RSVPD RSVPD Application D A T A Policy Control Admissions Control Routing Process Policy Control Admissions Control Packet Classifier Packet Scheduler DATA Packet Classifier Packet Scheduler DATA

Soft State Routers keep state about reservation Periodic messages refresh state, with PATH and RESV messages Non-refreshed state times out automatically Alternative: Hard state No periodic refresh messages. State is guaranteed to be there. State is kept till explicit removal. Properties of soft state: Adapts to changes in routes, sources, and receivers. Recovers from failures Cleans up state after receivers drop out

RSVP Reservation (1) R2 R3 1 2 PATH Host A 24.1.70.210 R1 3 R5 R4 PATH Host B 128.32.32.69 1. An application on Host A creates a session, 128.32.32.69/4078, by communicating with the RSVP daemon on Host A. 3. The PATH message follows the next hop path through R5 and R4 until it gets to Host B. Each router on the path creates soft session state with the reservation parameters. 2. The Host A RSVP daemon generates a PATH message that is sent to the next hop RSVP router, R1, in the direction of the session address, 128.32.32.69.

RSVP Reservation (2) R2 R3 PATH Host A 24.1.70.210 RESV R1 R5 6 R4 PATH 5 RESV 4 Host B 128.32.32.69 4. An application on Host B communicates with the local RSVP daemon and asks for a reservation in session 128.32.32.69/4078. The daemon checks for and finds existing session state. 6. The RESV message continues to follow the next hop path through R5 and R1 until it gets to Host A. Each router on the path makes a resource reservation. 5. The Host B RSVP daemon generates a RESV message that is sent to the next hop RSVP router, R4, in the direction of the source address, 24.1.70.210.

RSVP Multicast Reservation (1) Sender PATH PATH R1 PATH PATH R2 R3 PATH PATH Receiver PATH R4 PATH R5 R6 R7 PATH

RSVP Multicast Reservation (2) Sender R1 R2 R3 R4 R5 R6 R7 Receiver

Reservation Merging (3) 50Kbs (7) 100 Kbs Reservations merge as they travel up tree. R1 (6) 100 Kbs (2) 50Kbs R3 (9) 60Kbs (5) 100 Kbs R4 R6 R7 (1) 50Kbs (8) 60Kbs (4) 100 Kbs Receiver #1 Receiver #2 Receiver #3

RSVP: simple audio conference H1, H2, H3, H4, H5 both senders and receivers multicast group m1 no filtering: packets from any sender forwarded audio rate: b only one multicast routing tree possible H2 H3 H1 R1 R2 R3 H5 H4

RSVP: building up path state H1,, H5 all send path messages on m1: (address=m1, Tspec=b, filter-spec=no-filter,refresh=100) Suppose H1 sends first path message m1: in out L1 L2 L6 m1: in out L5 L6 L7 m1: in out L3 L4 L7 H2 H1 L2 L1 L6 R1 R2 R3 L5 L7 L4 L3 H3 H4 H5

RSVP: building up path state next, H5 sends path message, creating more state in routers m1: in out L1 L1 L6 L2 L6 m1: in out L5 L5 L6 L6 L7 m1: in out L3 L4 L7 H2 H1 L2 L1 L6 R1 R2 R3 L5 L7 L4 L3 H3 H4 H5

RSVP: building up path state H2, H3, H5 send path msgs, completing path state tables m1: in out L1 L1 L2 L6 L2 L6 m1: in out L5 L5 L6 L6 L7 L7 m1: in out L3 L4 L3 L4 L7 L7 H2 H1 L2 L1 L6 R1 R2 R3 L5 L7 L4 L3 H3 H4 H5

reservation msgs: receiver-to-network signaling reservation message contents: desired bandwidth filter type: no filter: any packets address to multicast group can use reservation fixed filter: only packets from specific set of senders can use reservation dynamic filter: senders who s packets can be forwarded across link will change (by receiver choce) over time. filter spec reservations flow upstream from receiver-to-senders, reserving resources, creating additional, receiverrelated state at routers

RSVP: receiver reservation example 1 H1 wants to receive audio from all other senders H1 reservation msg flows uptree to sources H1 only reserves enough bandwidth for 1 audio stream reservation is of type no filter any sender can use reserved bandwidth H2 H1 L2 L1 L6 R1 R2 R3 L5 L7 L4 L3 H3 H4 H5

RSVP: receiver reservation example 1 H1 reservation msgs flows uptree to sources routers, hosts reserve bandwidth b needed on downstream links towards H1 m1: in out L1 L2 L1 (b) L2 L6 L6 m1: in out L5 L5 L6 L6 (b) L7 L7 m1: in out L3 L3 L4 L4 L7 L7 (b) H2 H1 b L2 b L1 b L6 R1 R2 R3 b L5 b L7 b L3 b L4 H3 H4 H5

RSVP: receiver reservation example 1 (more) next, H2 makes no-filter reservation for bandwidth b H2 forwards to R1, R1 forwards to H1 and R2 (?) R2 takes no action, since b already reserved on L6 m1: in out L1 L1 (b) L2 L2 (b) L6 L6 m1: in out L5 L5 L6 L6 (b) L7 L7 m1: in out L3 L3 L4 L4 L7 L7 (b) H2 H1 b b L2 b b L1 b L6 R1 R2 R3 b L5 b L7 b L3 b L4 H3 H4 H5

RSVP: receiver reservation: issues What if multiple senders (e.g., H3, H4, H5) over link (e.g., L6)? arbitrary interleaving of packets L6 flow policed by leaky bucket: if H3+H4+H5 sending rate exceeds b, packet loss will occur m1: in out L1 L1 (b) L2 L2 (b) L6 L6 m1: in out L5 L5 L6 L6 (b) L7 L7 m1: in out L3 L3 L4 L4 L7 L7 (b) H2 H1 b b L2 b b L1 b L6 R1 R2 R3 b L5 b L7 b L3 b L4 H3 H4 H5

RSVP: example 2 H1, H4 are only senders send path messages as before, indicating filtered reservation Routers store upstream senders for each upstream link H2 will want to receive from H4 (only) H2 H3 L2 L3 H1 L1 L6 L7 R1 R2 R3 L4 H4

RSVP: example 2 H1, H4 are only senders send path messages as before, indicating filtered reservation in out L1, L6 L2(H1-via-H1 ; H4-via-R2 ) L6(H1-via-H1 ) L1(H4-via-R2 ) in out L4, L7 L3(H4-via-H4 ; H1-via-R3 ) L4(H1-via-R2 ) L7(H4-via-H4 ) H2 L2 R2 L3 H3 H1 L1 R1 L6 in L6, L7 L7 R3 L4 H4 out L6(H4-via-R3 ) L7(H1-via-R1 )

RSVP: example 2 receiver H2 sends reservation message for source H4 at bandwidth b propagated upstream towards H4, reserving b in out L1, L6 (b) L2(H1-via-H1 ;H4-via-R2 ) L6(H1-via-H1 ) L1(H4-via-R2 ) in out L4, L7 L3(H4-via-H4 ; H1-via-R2 ) L4(H1-via-62 ) L7(H4-via-H4 ) (b) H2 H1 b L2 L1 R1 b L6 in out R2 b L7 L6, L7 (b) L6(H4-via-R3 ) L7(H1-via-R1 ) R3 L3 b L4 H3 H4

RSVP: soft-state senders periodically resend path msgs to refresh (maintain) state receivers periodically resend resv msgs to refresh (maintain) state path and resv msgs have TTL field, specifying refresh interval in out L1, L6 (b) L2(H1-via-H1 ;H4-via-R2 ) L6(H1-via-H1 ) L1(H4-via-R2 ) in out L4, L7 L3(H4-via-H4 ; H1-via-R3 ) L4(H1-via-62 ) L7(H4-via-H4 ) (b) H2 H1 b L2 L1 R1 b L6 in out R2 b L7 L6, L7 (b) L6(H4-via-R3 ) L7(H1-via-R1 ) R3 L3 b L4 H3 H4

RSVP: soft-state suppose H4 (sender) leaves without performing teardown eventually state in routers will timeout and disappear! in out L1, L6 (b) L2(H1-via-H1 ;H4-via-R2 ) L6(H1-via-H1 ) L1(H4-via-R2 ) in out L4, L7 L3(H4-via-H4 ; H1-via-R3 ) L4(H1-via-62 ) L7(H4-via-H4 ) (b) H2 H1 b L2 L1 R1 b L6 in out R2 b L7 L6, L7 (b) L6(H4-via-R3 ) L7(H1-via-R1 ) R3 L3 b L4 H3 gone H4 fishing!

The many uses of reservation/path refresh recover from an earlier lost refresh message expected time until refresh received must be longer than timeout interval! (short timer interval desired) Handle receiver/sender that goes away without teardown Sender/receiver state will timeout and disappear Reservation refreshes will cause new reservations to be made to a receiver from a sender who has joined since receivers last reservation refresh E.g., in previous example, H1 is only receiver, H3 only sender. Path/reservation messages complete, data flows H4 joins as sender, nothing happens until H3 refreshes reservation, causing R3 to forward reservation to H4, which allocates bandwidth

RSVP scalability problems RSVP per-flow reservation model and soft-state philosophy are particularly suitable for multicast broadband applications (e.g. videoconference and video broadcasting) When used for point-to-point narrowband purposes (e.g. IP telephony) these choices implies large processing overhead in routers and great amount of traffic generation for periodic refreshes Example: ADPCM coding requires 32 kb/s for a voice channel. Neglecting packet overhead, a single OC-12 interface of a backbone router (622 Mb/s) should support up to 20000 flows, implying that: packet scheduler has to manage 20000 queues up to 20000 states must be periodically refreshed

RSVP scalability problems Solutions: Flows aggregation (SRP- Scalable Reservation Protocol) Use of RSVP limited to the Access Network (IP based), with interconnection among IP domains relying on other QoS capable technologies (e.g. ATM) Differentiated Services Approach Combination of RSVP/IntServ in the Access Section and DiffServ in the backbone Internet

IETF Differentiated Services Concerns with Intserv: Scalability: signaling, maintaining per-flow router state difficult with large number of flows Flexible Service Models: Intserv has only two classes. Also want qualitative service classes behaves like a wire relative service distinction: Platinum, Gold, Silver Diffserv approach: simple functions in network core, relatively complex functions at edge routers (or hosts)

Diffserv Architecture Edge router: per-flow traffic management marks packets as in-profile and out-profile b marking r scheduling. Core router: per class traffic management buffering and scheduling based on marking at edge preference given to in-profile packets

Edge-router Packet Marking profile: pre-negotiated rate A, bucket size B packet marking at edge based on per-flow profile Rate A B User packets Possible usage of marking: class-based marking: packets of different classes marked differently intra-class marking: conforming portion of flow marked differently than non-conforming one

Classification and Conditioning Packet is marked in the Type of Service (TOS) in IPv4, and Traffic Class in IPv6 6 bits used for Differentiated Service Code Point (DSCP) and determine PHB that the packet will receive 2 bits are currently unused

Classification and Conditioning may be desirable to limit traffic injection rate of some class: user declares traffic profile (e.g., rate, burst size) traffic metered, shaped if non-conforming

Core routers forwarding Routers define packet classes and separate incoming packets into classes. Treatment is done per class. Per-hop behavior (PHB) defines differences in performance among classes. PHB results in a different observable (measurable) forwarding performance behavior PHB does not specify what mechanisms to use to ensure required PHB performance behavior

Forwarding (PHB) PHBs developed: Expedited Forwarding: pkt departure rate of a class equals or exceeds specified rate logical link with a minimum guaranteed rate Providing low loss, low latency, low jitter, assured bandwidth, end-to-end service through DS domains Implies isolation: guarantee for the EF traffic should not be influenced by the other traffic classes Non-conformant traffic is dropped or shaped. Possible service: providing a virtual wire Assured Forwarding: 4 classes of traffic each guaranteed minimum amount of bandwidth each with three drop preference partitions

Forwarding (PHB) PHBs developed: Assured Forwarding: The intent is that it will be used to implement services that differ relative to each other (e.g., gold, silver, ). AF defines 4 classes with some bandwidth and buffers allocated to them. Each guaranteed minimum amount of bandwidth Each with three drop preference partitions Within each class, there are three drop priorities, which affect which packets will get dropped first if there is congestion. Non-conformant traffic is remarked.

AF table The DSCP (6 bit) pattern is: xyzab0 xyz is the class: 001-class1 ; 010-class2 ; 011-class3 ; 100-class4 ab is the drop precedence: 01-low ; 10-medium ; 11-high Class Class 1 Class 2 Class 3 Class 4 Drop precedence Low Drop 001010 (AF11) 010010 (AF21) 011010 (AF31) 100010 (AF41) Medium Drop 001100 (AF12) 010100 (AF22) 011100 (AF32) 100100 (AF42) High Drop 001110 (AF13) 010110 (AF23) 011110 (AF33) 100110 (AF43)

Service A service describes the overall treatment of a customer s traffic within a DS domain. Customers see services, not PHBs. To support a service, many components must work together: Mapping of service to PHBs, traffic conditioning, network provisioning, PHB-based forwarding. Services in the DiffServ architecture is defined in the form of Service Level Agreement (SLA).

QoS Summary The brute force approach has many supporters

Conventional IP Networks & Routing Client networks are connected to backbone via edge routers LAN, PSTN, ADSL Data packets are routed based on IP address and other information in the header Functional components Forwarding responsible for actual forwarding across a router consists of set of procedures to make forwarding decisions Control responsible for construction and maintenance of the forwarding table consists of routing protocols such as OSPF, BGP and PIM

Need for Multiprotocol Label Switching (MPLS) Forwarding function of a conventional router a capacity demanding procedure constitutes a bottle neck with increase in line speed MPLS simplifies forwarding function by taking a totally different approach by introducing a connection oriented mechanism inside the connectionless IP networks IP Router MPLS ATM Switch Control: IP Router Software Forwarding: Longest-match Lookup Control: IP Router Software Forwarding: Label Swapping Control: ATM Forum Software Forwarding: Label Swapping

Label Switching Decomposition of network layer routing into control and forwarding components applicable Label switching forwarding component algorithm uses forwarding table label carried in the packet What is a Label? Short fixed length entity

Label Switching Have a friend go to B ahead of you selecting the appropriate path. At every road they reserve a lane just for you. At ever intersection they post a big sign that says for a given lane which way to turn and what new lane to take. LANE#1 LANE#1 TURN RIGHT USE LANE#2 LANE#2

MPLS and ISO model TCP Applications UDP 7 to 5 IP 3 Frame MPLS PPP ATM (*) PPP Ethernet Relay ATM 2 Physical (Optical - Electrical) 1 4

MPLS Shim Header The Label (Shim Header) is represented as a sequence of Label Stack Entry Each Label Stack Entry is coded by 4 bytes (32 bits) as described 20 Bits is reserved for the Label Identifier (also named Label) Label (20 bits) Exp (3 bits) S (1 bit) TTL (8bits) Label : Label value (0 to 15 are reserved for special use) Exp : Experimental Use S : Bottom of Stack (set to 1 for the last entry in the label) TTL :Time To Live

Label Values 0-15 Reserved LABEL DESIGNATION 0 IPv4 Explicit Null 1 Router Alert 2 IPv6 Explicit Null 3 Implicit Null 4-14 Reserved for Future Use 15 OAM 16-2 20-1 Production Use

Forwarding Equivalence Class An MPLS capable router is called Label Edge Router (LER) if it resides at the edge of an MPLS network and Label Switching Router (LSR) if it resides in the core on an MPLS network. Forwarding Equivalence Class (FEC): A subset of packets that are all treated the same way by MPLS capable routers. A packet is assigned to an FEC at the ingress of an MPLS domain A packet s FEC can be determined by one or more of the following: Source and/or destination IP address Source and/or destination port number Protocol ID Differentiated services code point Incoming interface A particular PHB (scheduling and discard policy) can be defined for a given FEC

Forwarding Equivalence Classes LER LSR LSR LER LSP IP1 IP1 #L1 IP1 #L2 IP1 #L3 IP1 IP2 IP2 #L1 IP2 #L2 IP2 #L3 IP2 Packets are destined for different address prefixes, but can be mapped to common path FEC = A subset of packets that are all treated the same way by a router The concept of FECs provides for a great deal of flexibility and scalability In conventional routing, a packet is assigned to a FEC at each hop (i.e. L3 look-up), in MPLS it is only done once at the network ingress

MPLS Operation 1a. Routing protocols (e.g. OSPF-TE, IS-IS-TE) exchange reachability to destination networks 1b. Label Distribution Protocol (LDP) establishes label mappings to destination network 4. LER at egress removes label and delivers packet IP IP 2. Ingress LER receives packet and label s packets 3. LSR forwards packets using label swapping

MPLS Operation At ingress LER of an MPLS domain, an MPLS header is inserted to a packet before the packet is forwarded Label in the MPLS header encodes the packet s FEC At subsequent LSRs The label is used as an index into a forwarding table that specifies the next hop and a new label. The old label is replaced with the new label, and the packet is forwarded to the next hop. Egress LER strips the label and forwards the packet to final destination based on the IP packet header

Label Switched Path For each FEC, a specific path called Label Switched Path (LSP) is assigned The LSP is unidirectional To set up an LSP, each LSR must Assign an incoming label to the LSP for the corresponding FEC Labels have only local significance Inform the upstream node of the assigned label Learn the label that the downstream node has assigned to the LSP Need a label distribution protocol so that an LSR can inform others of the label/fec bindings it has made A forwarding table is constructed as the result of label distribution.

LSP Route Selection Hop-by-hop routing: use the route determined by the dynamic routing protocol Explicit routing (ER): the sender LSR can specify an explicit route for the LSP Explicit route can be selected ahead of time or dynamically Advantages Can establish LSP s based on policy, QoS, etc. Can have pre-established LSP s that can be used in case of failures. Signaling protocols CR-LDP RSVP-TE

MPLS BUILT ON STANDARD IP Dest Out 47.33 Dest Out 47.1 1 47.2 2 47.3 3 1 3 Dest Out 47.1 1 47.2 2 47.3 3 1 2 3 47.1 1 47.2 2 47.3 3 1 2 47.1 47.2 2 Destination based forwarding tables as built by OSPF, IS-IS, RIP, etc.

IP FORWARDING USED BY HOP- BY-HOP CONTROL 47.33 IP 47.1.1.1 Dest Out 47.1 1 47.2 2 47.3 3 3 Dest Out 47.1 1 47.2 2 47.3 3 IP 47.1.1.1 1 2 1 2 IP 47.1.1.1 Dest Out 47.1 1 47.2 2 47.3 3 1 2 47.1 IP 47.1.1.1 47.2

Label Distribution Intf Label Intf Label In In Out Out 3 50 1 40 Intf Label Intf In In Out 3 40 1 Request: 47.1 Dest Intf Label Out Out 3 47.1 1 50 1 1 Mapping: 40 2 47.3 3 2 3 2 1 47.1 47.2

Label Switched Path (LSP) Intf Label Dest Intf Label In In Out Out 3 0.50 47.1 1 0.40 Intf Label Dest Intf In In Out 3 0.40 47.1 1 Intf Dest Intf Label In Out Out 3 47.1 1 0.50 47.3 3 IP 47.1.1.1 1 2 3 2 1 3 IP 47.1.1.1 1 47.1 2 47.2

EXPLICITLY ROUTED LSP ER-LSP Intf In Dest Intf Out Label Out 3 47.1.1 2 1.33 3 47.1 1 0.50 47.3 3 IP 47.1.1.1 1 2 Intf Label Dest Intf Label In In Out Out 3 0.50 47.1 1 0.40 3 2 1 Intf Label Dest Intf In In Out 3 0.40 47.1 1 3 IP 47.1.1.1 1 47.1 2 47.2

Label Stacking A packet may carry multiple labels, organized as a last-in-first-out stack A label may be added to/removed from the stack at any LSR Processing always done on the top label Allow the aggregation of LSPs into a single LSP for a portion of the route, creating a tunnel It allows LSPs to be tunneled in other LSPs At the beginning of the tunnel, the LSR assigns the same label to packets from different LSPs by pushing the label onto each packet s stack At the end of the tunnel, the LSR pops the top label

Label Stacking

Implicit Null