Mul$media Networking. #5 Real- Time Transport Protocol Semester Ganjil 2012 PTIIK Universitas Brawijaya
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1 Mul$media Networking #5 Real- Time Transport Protocol Semester Ganjil 2012 PTIIK Universitas Brawijaya
2 Schedule of Class Mee$ng 1. Introduc$on 2. Applica$ons of MN 3. Requirements of MN 4. Coding and Compression 5. RTP 6. IP Mul$cast 7. IP Mul$cast (cont d) 8. Overlay Mul$cast 9. CDN: Solu$ons 10. CDN: Case Studies 11. QoS on the Internet: Constraints 12. QoS on the Internet: Solu$ons 13. Discussion 14. Summary
3 Today s Outline RTP protocol goals mixers and translators control: awareness, QOS feedback media adapta$on
4 Real- Time Protocol (RTP) End- to- end protocol for applica$ons transmi_ng real- $me data, such as audio and video RFC 3550 RTP packets encapsulated in UDP segments RTP encapsula$on only seen at end systems (not by intermediate routers) routers provide best- effort service, no special effort to ensure RTP packets arrive at des$na$on in $mely manner
5 RTP: Big Picture Applica$on Media Encapsula$on RTP RTCP data UDP control ST- II IPv4/6 Ethernet AAL5 ATM
6 RTP: Big Picture Real- Time Transport Protocol (RTP) = data + control data: $ming, loss detec$on, content labeling, talk spurts, encryp$on control: (RTCP) periodic with T popula$on QOS feedback membership es$ma$on loop detec$on
7 RTP: Goals lightweight: specifica$on and implementa$on flexible: provide mechanism, don t dictate algorithms protocol- neutral: UDP/IP,ST- II,IPX,ATM- AALx,... scalable: unicast, mul$cast from 2 to O(107) separate control/data: some func$ons may be taken over by conference control protocol secure: support for encryp$on, possibly authen$ca$on
8 RTP runs on top of UDP Applica$on layer protocol RTP libraries provide transport- layer interface that extends UDP: port numbers, IP addresses payload type iden$fica$on packet sequence numbering $me- stamping Applica$ons that use RTP are: Less sensi$ve to packet loss Very sensi$ve to packet delays UDP provides key services: Mul$plexing Checksum
9 RTP Func$ons segmenta$on/reassembly done by UDP (or similar) re- sequencing (if needed) loss detec$on for quality es$ma$on, recovery intra- media synchroniza$on: remove delay jiker through playout buffer intra- media synchroniza$on: driling sampling clocks inter- media synchroniza$on (lip sync between audio and video) quality- of- service feedback and rate adapta$on source iden$fica$on
10 RTP header bit V(2) P X CSRC count Mpayload type timestamp sequence number UDP packet synchronization source identifier (SSRC) contributing source identifiers (CSRC) header extension payload (audio,video,...) 0x00 opt. opt. opt. bytes
11 RTP header payload type sequence number type time stamp Synchronization Source ID Miscellaneous fields payload type (7 bits): audio/video encoding method; may change during session. If sender changes encoding during call, sender informs receiver via 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: Mo$on JPEG Payload type 31: H.261 Payload type 33: MPEG2 video sequence # (16 bits): +1 for each RTP packet sent detect packet loss, restore packet sequence
12 RTP header payload type sequence number type time stamp Synchronization Source ID Miscellaneous fields *mestamp field (32 bits long): sampling instant of first byte in this RTP data packet for audio, $mestamp clock increments by one for each sampling period (e.g., each 125 usecs for 8 KHz sampling clock) if applica$on generates chunks of 160 encoded samples, $mestamp increases by 160 for each RTP packet when source is ac$ve. Timestamp clock con$nues to increase at constant rate when source is inac$ve. SSRC field (32 bits long): synchroniza$on source sources pick at random may change aler collision!
13 RTP header payload type sequence number type time stamp Synchronization Source ID Miscellaneous fields P: padding (for encryp$on) last byte has padding count M: marker bit; frame, start of talk spurt delay adjustment CC: content source count (for mixers) CSRC: iden$fiers of those contribu$ng to (mixed into) packet
14 RTP example example: sending 64 kbps PCM- encoded voice over RTP applica$on collects encoded data in chunks, e.g., every 20 msec = 160 bytes in a chunk audio chunk + RTP header form RTP packet, which is encapsulated in UDP segment RTP header indicates type of audio encoding in each packet sender can change encoding during conference RTP header also contains sequence numbers, $mestamps
15 Today s Outline RTP protocol goals mixers and translators control: awareness, QOS feedback media adapta$on
16 RTP: Mixers and Translators mixer: several media stream one new stream (new encoding) reduced bandwidth networks (dial- up) appears as new source, with own iden$fier translator: single media stream may convert encoding protocol transla$on (na$ve ATM IP), firewall all packets: source address = translator address Goals: Accommodate par$cipant network resources
17 RTP: Mixers and Translators end system SSRC= end system SSRC= DVI4 L16 GSM mixer SSRC=5 translator GSM SSRC=5 CSRC=
18 Today s Outline RTP protocol goals mixers and translators control: awareness, QOS feedback media adapta$on
19 Control: awareness, QOS feedback Packet loss, conges$on, jiker, delivery $mes Directly useful for control of adap$ve encodings Iden$fy if problems are local or global Short- term and long- term sta$s$cal analysis Self- adjus$ng network Each par$cipant eventually knows about the other members Source descrip$on dynamically iden$fies who is sending Ac$ve senders get more bandwidth Session bandwidth kept constant by adjus$ng transmission rate based on the number of par$cipants
20 Real- Time Control Protocol (RTCP) works in conjunc$on with RTP each par$cipant in RTP session periodically sends RTCP control packets to all other par$cipants each RTCP packet contains sender and/or receiver reports report sta$s$cs useful to applica$on: # packets sent, # packets lost, interarrival jiker feedback used to control performance sender may modify its transmissions based on feedback
21 RTCP: packet types stackable packets, similar to data packets sender report (SR): bytes send es$mate rate; $mestamp synchroniza$on recep0on reports (RR): number of packets sent and expected loss, avg. inter- arrival jiker, round- trip delay source descrip0on(sdes): name, , loca$on,... CNAME (canonical name = user@host) iden$fies user across media explicit leave (BYE): in addi$on to $me- out extensions (APP): applica$on- specific (none yet)
22 RTCP: packet structure if encrypted: random 32-bit integer packet packet packet receiver reports item chunk item item chunk item SR SSRC sender report SSRC site 1 SSRC site 2 SDES SSRC CNAME PHONE CNAME LOC SSRC BYE SSRC SSRC reason compound packet UDP packet
23 RTCP: mul$ple mul$cast senders sender RTP RTCP RTCP RTCP receivers each RTP session: typically a single mul$cast address; every par$cipant: periodically mul$cast RTCP packet to same group as data RTP, RTCP packets dis$nguished from each other via dis$nct port numbers to limit traffic, each par$cipant reduces RTCP traffic as number of conference par$cipants increases
24 RTCP: announcement interval Goals: es$mate current number of par$cipants & iden$$es of par$cipants dynamic source descrip$on ( SDES ) who s talking? quality- of- service feedback adjust sender rate to O(1000) par$cipants, few % of data randomized response with rate as members group size limited by tolerable age of status gives ac$ve senders more bandwidth sol state: delete if silent
25 RTCP: bandwidth scaling RTCP aaempts to limit its traffic to 5% of session bandwidth example : one sender, sending video at 2 Mbps RTCP akempts to limit RTCP traffic to 100 Kbps RTCP gives 75% of rate to receivers; remaining 25% to sender 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. par$cipant determines RTCP packet transmission period by calcula$ng avg RTCP packet size (across en$re session) and dividing by allocated rate
26 sender period T : receivers: RTCP: bandwidth scaling T = T = # of senders session bw #ofreceivers session bw avg. RTCP packet size avg. RTCP packet size next packet = last packet + max(5 s, T ) random( ) randomization prevents bunching to reduce RTCP bandwidth, alternate between SDES components
27 RTCP sender reports (SR) SSRC of sender: iden$fies source of data NTP *mestamp: when report was sent RTP *mestamp: corresponding RTP $me lip sync sender s packet count: total number sent sender s octet count: total number sent followed by zero or more receiver report
28 RTCP receiver reports (RR) SSRC of source: iden$fies who s being reported on frac*on lost: binary frac$on cumula*ve number of packets lost: long- term loss highest sequence number received: compare losses, disconnect inter- arrival jiaer: smoothed inter- packet distor$on LSR: $me last SR heard DLSR: delay since last SR
29 RTCP: round trip delay es$ma$on compute round- trip delay between data sender and receiver [10 Nov :33:25.125] [10 Nov :33:36.5] n SR(n) A=0xb710:8000 ( s) ntp_sec =0xb44db705 ntp_frac=0x ( s) r RR(n) dlsr=0x ( s) lsr =0xb705:2000 ( s) DLSR (5.25 s) A 0xb710:8000 ( s) DLSR 0x0005:4000 ( s) LSR 0xb705:2000 ( s) delay 0x 6:2000 ( s)
30 RTCP: stream synchroniza$on RTCP can synchronize different media streams within a RTP session e.g., videoconferencing app: each sender generates one RTP stream for video, one for audio. $mestamps in RTP packets $ed to the video, audio sampling clocks not $ed to wall- clock $me each RTCP sender- report packet contains (for most recently generated packet in associated RTP stream): $mestamp of RTP packet wall- clock $me for when packet was created receivers uses associa$on to synchronize playout of audio, video
31 RTCP: stream synchroniza$on = sync different streams (audio, video, slides,... ) $mestamps are offset with random intervals may not $ck at nominal rate SRs correlate real $me (wall clock $me) with RTP ts 560 = 8:45:17.23 audio RTP RTCP SR RTP timestamp 1800 = 8:45:17.18 video RTCP SR
32 Today s Outline RTP protocol goals mixers and translators control: awareness, QOS feedback media adapta$on
33 Media Adapta$on Mul$media applica$ons can adjust their data rates: Audio: (MPEG L3), encoding, sampling rate, mono/ stereo encoding sampling rate bit rate LPC 8,000 5,600 GSM 8,000 13,200 DVI4 8,000 32,000 µ-law 8,000 64,000 DVI4 16,000 64,000 a range of DVI4 and MPEG L3 L16 stereo 44,100 1,411,200
34 Media Adapta$on Video: frame rate, quan$za$on, image resolu$on, encoding Size [Bytes] Noise Face 1 Face 2 White Picture Black Picture Q-Factor
35 Applica$on Control networks without QoS or shared reserved link: adapt applica$on to available bandwidth share bandwidth fairly with TCP? lowest common denominator mixers, translators
36 RTP/RTCP Does NOT Define media data formats or encodings Need media specific profiles Handle connec$on setups Need other protocols like SIP or H.323 Handle resource reserva$on Need other protocols like RSVP Guarantee $mely data delivery or Quality of Service However, it does provide necessary data to applica$on to order packets and adjust signal quality
37 References RFC hkp://tools.ie{.org/html/rfc1889 RFC hkp://tools.ie{.org/html/rfc3551 Wikipedia: RTP - hkp://en.wikipedia.org/wiki/real- $me_transport_protocol RTCP - hkp://en.wikipedia.org/wiki/rtcp 10/2/12 Real- $me Transport Protocol 37
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