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Transcript
Media: Voice and Video in your SIP Environment Jitendra Shekhawat Agenda Objective: Introduction of Media in the SIP environment. • • • • Common Audio and Video Codecs Media/Codec Negotiations Tuning Your Network for Voice and Video QoS issues, metrics and user quality expectations IP Audio/Video Telephony Network •Call Control: SIP SIP Video Endpoints •Media: RTP SIP Soft Phone •Video: H263, H264, MPEG4 •Audio: G711, G723, G729, G726, AMR-NB, etc. SIP Desk Phone SIP SIP RTP Multimedia Server IP RTP PC – Email Client SIP SIP RTSP RTP RTP Broadband Users SIP Proxy Server Applications RTSP Streaming Server CNN, ESPN, Bloomberg, live feed • Video Mail • Video Portal • Live content streaming SIP Call Example Audio Video Codecs and Payload Types • RFC 3551 • Some codecs PT 0 3 4 8 9 18 34 Dynamic Dynamic Dynamic Dynamic encoding name pcmu gsm g723 pcma g722 g729 h263 iLBC AMR AMR-WB AMR-WB+ Media Transport • RTP – Real Time Transport Protocol – media packet transport – One stream per direction between endpoints • RTCP – RTP Control Protocol – Provides quality information – Generate reports to the network RTP Packet RTP Datagram IP Header 20 bytes Version 2 bits Padding 1 bit RTP Datagram UDP Header 8 bytes Extensio n 1 bit RTP Header 12 bytes CSRC count 4 bits Marker 1 bit RTP Datagram RTP Payload N bytes Payload Type 7 bits Sequence Number 2 bytes Time stamp 4 bytes Source Identifier 4 bytes RTCP Packet • Receiver of RTP stream sends periodic updates to the originator • Packet count • Byte count • Packet loss • Timestamps to assess round-trip delay • Jitter RTP Packet Payload size Function of: codec speed, frame-size Frequency packets sent Payload size = codec speed X frame size 8 X 1000 bits/byte msec / sec Example: g.711, 20 ms frames: 64000 bps X 20 msec / 8 = 160 byte payload Media Stream (RTP) Bandwidth: Packet size := Header + Payload Header := Ethernet + (IP + UDP + RTP) = 38 + (20 + 8 + 12) = 38 + 40 bytes Payload := depends on codec Example: g.711, 20 ms frames (50 packets/s) 160 byte payload + (38 + 40) byte header IP bandwidth: 200 byte/packet = 80,000 bps 160 kbps for 2 way Ethernet bandwidth: 238 byte/packet = 95,2000 bps 190.4 kbps for 2 way • Ethernet: Preamble (8) + Ethernet Header (14) + Ethernet CRC (4) + Inter-frame gap (12) = 38 Codec Bandwidths Coder Bitrate Encoded bandwidth G.711 64 kbps 200-3400 Hz G.723 5.4 or 6.3 kbps 200-3400 Hz G.729A (20ms Packet) 8 kbps 200-3400 Hz AMR 4.75 to 12.2 kbps 200-3400 Hz AMR-WB Variable: 6.6 up to 23.85 (noncontinuous) 50 to 7000 Hz AMR-WB+ Variable: 6-36 kbps (mono) or 748 kbps (stereo) 50 Hz – 7.2 kHz up to 50 Hz – 19.2 kHz iLBC 13.33 kbps for 30 ms, 15.20 kbps for 20ms 200-3400 Hz Codec Bandwidths Coder IP Bandwidth / RTP stream G.711 (30 ms Packet) 74.6 kbps G.711 (20ms Packet) 80 kbps G.711 (10 ms Packet) 96 kbps G.723.1 (30ms Packet) 15.7 kbps G.729A (20ms Packet) 24 kbps AMR (20 ms) 20.4 - 28 kbps AMR-WB (20ms) 22.4 – 39.6 kbps AMR-WB+ (20ms) 22 – 52 kbps iLBC (20ms or 30ms) 31.2 kbps or 24 kbps Video streams I-frames (Key frames) P-frames (predicted frames) Frame Sequence 10000 9000 8000 Frame Size 7000 6000 5000 4000 3000 2000 1000 0 1 18 35 52 69 86 103 120 137 154 171 188 205 222 239 256 273 290 307 324 341 358 375 392 409 426 443 460 477 494 511 528 545 562 579 596 613 630 647 664 681 698 Frame Number Video Formats (IP vs. 3G) • High resolution for IP networks – More bandwidth available – SIP Video Phones are generally CIF size (352 × 288 pixels) – Recommended: CIF, 15 or 30fps, up to 384kbps • Low resolution for 3G networks – Total bandwidth limited to 64kbps – Generally video + audio is approx 52kbps (12.2kbps AMR + 40kbps H263) – 3G Mobile phones are generally QCIF size (176 × 144 pixels) 4 CIF 3 QCIF Performance Issues • Propagation Delay Time required to travel end to end across the network • Transport Delay Time required to traverse network equipment • Packetization Delay Time to digitize, build frames and undo at destination • Jitter Delay Fixed delay by receiver to hold 1 or more packets to damp variations in arrival times • Packet Loss Packet size impacts sound quality Jitter Delay • Calculated on inter-arrival time of successive packets – Average inter-arrival time – Standard deviation • Goal inter-arrival time = inter-arrival time on emitted packets • 3 phenomena effecting jitter – Packet loss (threshold 5%) – Silence suppression – Out of sequence packets • Can be configured on most VoIP equipment Packet Fragmentation • Audio RTP packets – Not generally fragmented since packet size is less than MTU • Video RTP packets – A large frame is fragmented into a series of packets for transmission over network – I-Frame fragmentation • Receiver must receive all fragments to properly reconstruct frame Packet Loss • Audio – Packet Loss Concealment (PLC) • Mask effect of lost or discarded packets • Replay previous packet or use previous packets to estimate missing data • Key method for improving voice quality – Packet Loss Recovery (PLR) – Packet Redundancy • Increased bandwidth • Video – I-Frame • If a fragment is lost, subsequent P-Frames will not be sufficient to reconstruct image at receiver • Video conversion tools available to generate files more suitable for real-time transmission G.107 to MOS mapping Codec Bandwidth and Voice Quality Comparison Codec Payload Bit Rate Voice Quality G.711 64 Kbps Excellent (MOS 4.2) G.723 6.4 Kbps / 5.3 Kbps Good (MOS 3.9) Fair (MOS 3.7) G.729 8 Kbps Good (MOS 4.0) G.726 or G.721 16/24/32/40 Kbps 2/3.2/4/4.2 iLBC 13.33/15.2 kbps Good (MOS 4.0) AMR-WB+ 6-36 kbps Good (MOS near 4.0) Network Issues? Network Issues – Now What • Determine the source of delay – Codec’s? – Too many hops? – Not enough bandwidth? • Define means to reduce delay – Provision smaller packet sizes – Reduce hop count – Change routing protocols used • Keep monitoring – Find problems first – Objectively identify issues IP Header Traffic Shaping • DiffServ • RSVP • MPLS Conclusion • Reliability – Can calls be made when needed? – Will call setup time match current environment? – Will calls be disconnected? • Quality – Is the voice quality of the calls the same? – Can the users tell the difference? • Cost – What are the cost benefits of VoIP? – What equipment will be needed? Wrap-up Q & A / Quiz Frame Sizes Format Sub-QCIF (SQCIF) Dimension (H x W, pixels) >1 bits/pixel 128 x 96 Quarter-CIF (QCIF) 176 x 144 CIF (Common Intermediate Format) 352 x 288 4CIF (4 x CIF) 704 x 576 16CIF (16 x CIF) 1408 x 1152