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Media: Voice and Video in your SIP Environment PowerPoint Presentation
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Media: Voice and Video in your SIP Environment

Media: Voice and Video in your 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