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Media: Voice and Video in your SIP Environment

Learn about common audio and video codecs, media negotiations, network tuning for voice and video, QoS issues, metrics, and user quality expectations in the SIP environment.

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Media: Voice and Video in your SIP Environment

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  1. Media: Voice and Video in your SIP Environment Jitendra Shekhawat

  2. Agenda • Common Audio and Video Codecs • Media/Codec Negotiations • Tuning Your Network for Voice and Video • QoS issues, metrics and user quality expectations Objective: Introduction of Media in the SIP environment.

  3. IP RTSP Streaming Server SIP Proxy Server IP Audio/Video Telephony Network • Call Control: SIP • Media: RTP • Video: H263, H264, MPEG4 • Audio: G711, G723, G729, G726, AMR-NB, etc. SIP Video Endpoints SIP Soft Phone SIP Desk Phone SIP SIP RTP RTP PC – Email Client Multimedia Server SIP SIP Broadband Users RTSP RTP RTP Applications • Video Mail • Video Portal • Live content streaming CNN, ESPN, Bloomberg, live feed

  4. SIP Call Example

  5. Audio Video Codecs and Payload Types • RFC 3551 • Some codecs

  6. 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

  7. RTP Packet RTP Datagram RTP Datagram RTP Datagram IP Header 20 bytes UDP Header 8 bytes RTP Header 12 bytes RTP Payload N bytes Version 2 bits Padding 1 bit Extension 1 bit CSRC count 4 bits Marker 1 bit Payload Type 7 bits Sequence Number 2 bytes Time stamp 4 bytes Source Identifier 4 bytes

  8. 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

  9. RTP Packet Payload size Function of: codec speed, frame-size Example: g.711, 20 ms frames: 64000 bps X 20 msec / 8 = 160 byte payload Frequency packets sent codec speed X frame size Payload size = 8 X 1000 bits/byte msec / sec

  10. 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

  11. Codec Bandwidths

  12. Codec Bandwidths

  13. Video streams I-frames (Key frames) P-frames (predicted frames) Frame Sequence

  14. 4 CIF 3 QCIF 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)

  15. 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

  16. 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

  17. 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

  18. 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

  19. G.107 to MOS mapping

  20. Codec Bandwidth and Voice Quality Comparison

  21. Network Issues?

  22. 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

  23. IP Header

  24. Traffic Shaping • DiffServ • RSVP • MPLS

  25. 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?

  26. Wrap-up Q & A / Quiz

  27. Frame Sizes

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