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Archive for the ‘TelePresence’ Category

EX60 Recorded via TCS

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TelePresence training this week continues.  I’m learning a ton, here at the CDW Minneapolis office with @benpolzin and @Dixon4UK.  Here’s quick EX60 video recorded through TCS.  The quality isn’t superb…still some settings to tweak.

Written by Matthew Berry

March 1st, 2012 at 10:18 am

Posted in TelePresence

TelePresence Training

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Rocking it out on as EX60!

20120227-103653.jpg

Written by Matthew Berry

February 27th, 2012 at 9:37 am

Posted in TelePresence,Updates

Cisco TelePresence Fundamentals 05

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Continuing the TelePresence notes series based on Cisco TelePresence Fundamentals by Cisco Press.

TelePresence Network Interaction

As highlighted in each TelePresence system connectivity schematic, the primary codec is the interface between the CTS system and the network infrastructure. The primary codec connects to the network access-edge switch through an RJ-45 10/100/1000 port. The access-edge Catalyst switch that it connects to provides IP services, 802.1Q/p VLAN serv- ices, QoS services, and security services to the TelePresence system.

TelePresence systems use a private network for internal communications between the primary and secondary codecs and between codecs and cameras. By default, the internal address range is 192.168.0.0/24 through 192.168.4.0/24; however, if the TelePresence codec receives a 192.168.x.x address from the network, the internal private net- work switches to 10.0.0.0/24 through 10.0.4.0/24.

Even though only 192.168.0.0/24 through 192.168.3.0/24 are illustrated, 192.168.4.0/24 is reserved within the system for future (internal) use.

Similarly, if the TelePresence system uses 10.0.0.0/24 through 10.0.3.0/24 for its internal net- working address range, 10.0.4.0/24 is reserved within the system for future (internal) use.

Following are three key points regarding the internal networking of TelePresence systems:

  • From the network’s perspective, the TelePresence primary codec appears as a single endpoint device with a single IP address. (Remember, the 7975G IP Phone also appears as a separate endpoint device with its own IP address).
  • The internal components (such as secondary codecs and cameras) do not receive a default gateway; therefore, they cannot route beyond the primary codec.
  • If the primary codec uses 192.168.0.0/24 through 192.168.4.0/24 as its internal networking addresses (which is the default), it cannot connect to external servers or end-points that uses these same addresses (because it will attempt to reach such addresses via its internal network, not its external default gateway). Conversely, if the primary codec has been assigned an IP address from the network in the 192.168.x.x range, it uses internal networking addresses in the range of 10.0.0.0/24 through 10.0.4.0/24 and similarly cannot connect to external servers or endpoints that might use these same addresses.

Cisco TelePresence network control, management, and signaling protocol:

Cisco TelePresence signaling and media paths:

When the TelePresence system completes these protocol interactions, it is ready to place and receive calls. When a call initiates, the following steps occur:

  1. The Cisco 7975G IP Phone sends an XML Dial message to its primary codec.
  2. The initiating TelePresence primary codec forwards the request as a SIP Invite message to the CUCM.
  3. The CUCM, in turn, forwards the SIP Invite message to the destination TelePresence primary codec (or Session Border Controller, in the case of business-to-business calls).
  4. The destination codec forwards the message as an XML Ring message to its associated 7975G IP Phone. (The TelePresence primary codec can optionally be set to automatically answer the incoming call, in which case the codec answers the call immediately and proceeds to Step 6, which is to send a SIP OK message to CUCM.)
  5. If auto-answer is not enabled, when the user presses the Answer softkey on the 7975G IP Phone, the 7975G IP Phone replies with an XML Answer message to the destination codec.
  6. The destination codec sends a SIP 200 OK message to the CUCM.
  7. The CUCM relays this SIP 200 OK message to the initiating TelePresence primary codec, and the call is established.
  8. Real-time media, both audio and video, passes between the TelePresence primary codecs over Real Time Protocol (RTP).

 

Written by Matthew Berry

February 20th, 2012 at 2:01 pm

Posted in TelePresence

Cisco TelePresence Fundamentals 04

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Continuing the TelePresence notes series based on Cisco TelePresence Fundamentals by Cisco Press.

Managing Latency, Jitter, and Loss

A few important facts about latency:

  • Latency: The human experience is round-trip in nature. This is referred to as conversational latency, or experience-level latency; 250 ms to 350 ms is the threshold at which the human mind begins to perceive latency and be annoyed by it.
  • Latency Target: To maintain acceptable experience-level latency, Cisco recommends that customers engineer their networks with a target of no more than 150 ms of network-level latency, in each direction, between any two TelePresence systems.
  • Latency Thresholds: When network-level latency exceeds 250 ms averaged over any 10-second period, the Cisco TelePresence system receiving those packets generates an alarm, and an onscreen message displays to the user.

If the latency from one TelePresence system in Hong Kong to the CTMS in London is 125 ms, and the latency from the CTMS in London to the other TelePresence system in San Francisco is 125 ms, the end-to-end latency from the Ethernet network interface of the Hong Kong system to the Ethernet network interface of the San Francisco system is 250 ms, plus approximately 10 ms added by the CTMS, for a total of 260 ms. The TelePresence System in San Francisco will not realize this and will think that the latency for that meeting is only 125 ms.

A few important facts about jitter:

  • Jitter Target: To maintain acceptable experience-level latency, Cisco recommends that customers engineer their networks with a target of no more than 10 ms of packet-level jitter and no more than 50 ms of video frame jitter in each direction between any two TelePresence systems.
  • Jitter Thresholds: Cisco TelePresence uses a quasi-adaptive jitter buffer.
    • At the beginning of every new meeting, the jitter buffer starts out at 85 ms in depth.
    • After monitoring the arrival time of the video frames for the first few seconds of the meeting, if the incoming jitter exceeds 85 ms average, the jitter buffer is dynamically adjusted to 125 ms.
    • After that, if the jitter exceeds 125 ms averaged over any 10-second period, the Cisco TelePresence system receiving those video frames generates an alarm and dynamically adjusts the jitter buffer to 165 ms. The alarm is written to the syslog log file of that TelePresence system, and an SNMP trap message is generated.
    • No onscreen message is displayed to the user.Any packets exceeding the 165 ms jitter buffer depth are discarded by the receiving TelePresence system and logged as “late packets” in the call statistics.

A few important facts about loss:

  • Loss Target: To maintain acceptable experience-level video quality, Cisco recommends that customers engineer their networks with a target of no more than .05 percent packet loss in each direction between any two TelePresence systems. This is an incredibly small amount, and given the complexity of today’s global networks, 0.05 percent loss is not always possible to accomplish.
  • Loss Thresholds: The Cisco TelePresence system has a multi-tiered loss thresholds:
    • When packet loss (or late packets) exceeds 1 percent averaged over any 10-second period, the Cisco TelePresence system receiving those packets generates an alarm, and an onscreen message appears.
    • When packet loss (or late packets) exceeds 10 percent averaged over any 10-second period, the Cisco TelePresence system receiving those packets generates a second alarm, and a second on-screen message appears (unless the hold timer is already in affect).
    • If loss (or late packets) exceeds 10 percent averaged over any 60-second period, in addition to the actions described, the system downgrades the quality of its outgoing video.
    • Finally, if loss equals 100 percent for greater than 30 seconds, the codec hangs up the call. If the packets begin flowing again anytime up to the 30-second timer, the codec immediately recovers.

Written by Matthew Berry

February 15th, 2012 at 11:44 am

Posted in TelePresence

Cisco TelePresence Fundamentals 03

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Continuing the TelePresence notes series based on Cisco TelePresence Fundamentals by Cisco Press.

Frame Intervals and Encoding Techniques

Each 33-ms frame interval contains encoded slices of the video image. A reference frame is the largest and least compressed and contains a complete picture of the image. After sending a reference frame, subsequent frames contain only the changes in the image since the most recent reference frame.

Reference frames are sent at the beginning of a call, or anytime the video is interrupted, such as when the call is placed on hold and the video suspended, and then the call is taken off of hold and the video resumed.

An Instantaneous Decode Refresh (IDR) Frame is a reference frame containing a complete picture of the image. When an IDR frame is received, the decode buffer is refreshed so that all previously received frames are marked as “unused for reference,” and the IDR frame becomes the new reference picture. IDR frames are sent by the encoder at the beginning of the call and at periodic intervals to refresh all the receivers. They can also be requested at any time by any receiver.

There are two pertinent examples of when IDRs are requested by receivers:

  • In a multipoint meeting, as different sites speak, the Cisco TelePresence Multipoint Switch (CTMS) switches the video streams to display the active speaker. During this switch, the CTMS sends an IDR request to the speaking endpoint so that all receivers can receive a new IDR reference frame.
  • Whenever packet loss occurs on the network and the packets lost are substantial enough to cause a receiver to lose sync on the encoded video image, the receiver can request a new IDR frame so that it can sync back up.

A new technique, implemented at the time this book was authored, is Long-Term Reference (LTR) Frames. LTR allows for multiple frames to be marked for reference, providing the receiver with multiple points of reference to reconstruct the video image.

AAC-LD Compression Algorithm

Cisco TelePresence uses the latest audio encoding technology known as Advanced Audio Coding–Low Delay (AAC-LD). AAC is a wideband audio coding algorithm designed to be the successor of the MP3 format and is standardized by the International Organization for Standardization/International Electrotechnical Commission (ISO/IEC) Moving Picture Experts Group (MPEG).

AAC-LD (Low Delay) bridges the gap between the AAC codec, which is designed for high-fidelity applications such as music, and International Telecommunication Union (ITU) speech encoders such as G.711 and G.722, which are designed for speech. AAC-LD combines the advantages of high-fidelity encoding with the low delay necessary for realtime, bidirectional communications.

The AAC-LD standard allows for a wide range of sample frequencies (8 kHz to 96 kHz). Cisco TelePresence implements AAC-LD at 48 kHz sampling frequency. This means that the audio is sampled 48,000 times per second, per channel. These samples are then encoded and compressed to 64 kbps, per channel, resulting in a total bandwidth of 128 kbps for single-screen systems (two channels) and 256 kbps for multiscreen systems (four channels).

RTP Packet Format

It’s always good to look at an RTP packet to remember what it contains:

RTP Packet

Comparison of Voice and Video on the Network

Adding video to your network isn’t a matter of multiplying bandwidth requirements by two.  Video requires a significant amount of bandwidth and has a higher rate of activity per sample period, in the case below 33ms.

Voice vs. Video

Video traffic appears as a series of video frames spaced at regular intervals. In the case of Cisco TelePresence, video frames are sent approximately every 33 msec. The size of each frame varies based on the amount of changes since the previous frame. Therefore, the overall characteristic of TelePresence video is a relatively bursty, variable bit-rate stream.

A video frame can also be referred to as an Access Unit in H.264 terminology. The H.264 standard defines two layers, a Video Coding Layer (VCL) and a Network Abstraction Layer (NAL).

The figure below shows a simplified example of how TelePresence video is mapped into RTP packets:

TelePresence Packet Mapping

Each video frame consists of multiple RTP packets spaced out over the frame interval. The boundary of each video frame is indicated through the use of the marker bit.  Each RTP packet contains one or more NAL Units (NALU), depending upon the packet type: single NAL unit packet, single-time or multi-time aggregation packet, or fragmentation unit (part of a NALU). Each NALU consists of an integer number of bytes of coded video.

RTP Packets within a single video frame and across multiple frames are not necessarily independent of each other. In other words, if one packet within a video frame is discarded, it affects the quality of the entire video frame and might possibly affect the quality of other video frames.

Written by Matthew Berry

February 14th, 2012 at 11:32 am

Posted in TelePresence