Download Media: Voice and Video in your SIP Environment

Media: Voice and Video in
your SIP Environment
Jitendra Shekhawat
Agenda
Objective: Introduction of Media in the SIP environment.
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•
•
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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