Telepresence systems based on IMS networks
What is Telepresence?
Essentially, telepresence allows people to be effectively present without the hassle of travel. A conference call is a typical form of virtual presence: whether it is an audio or video conference, it gives the feeling of presence to the participants. The quality of presence differs based on how much information is provided to the participants. If it is only an audio connection, the presence awareness is very small as participants do not have visual contact. Visual contact is, of course, present in video conferences, but participants still do not have a strong sense of realism, which requires the extra dimensions of eye contact, body language, spatial audio to distinguish direction of voice, and natural image size to simulate a life-size image of the participating person. A telepresence system incorporating all these aspects offers conference communications with a high audio-visual experience, giving the participants a strong sense of presence and the feeling that all participants are sitting at the same room around the same table.
A telepresence system consists of dedicated hardware and software, linked using sufficient bandwidth to provide high throughput of HD audio and video information from multiple items of equipment. Participants appear in life size on large TV screens; HD cameras are capturing video from different angles; and microphones are recording audio from different positions. Robotic arms with cameras are also used for tracking different people and places to capture different additional objects like whiteboards, demonstration equipment or additional audience. Furniture and lighting conditions in telepresence rooms provide an environment where conference calls simulate a real-life environment and participants lose the feeling of multiple physical locations.
Telepresence networks
Current telepresence systems are based on an IP network which connects at least two end-points. The main precondition for successful telepresence is an IP based connection with a bandwidth sufficient enough to receive and send multimedia data in high definition. The amount of data depends on the quality of video and audio in a telepresence session. For example, a telepresence system such as the Cisco CTS-3000 with triple screen video of 1080 pixels resolution requires a 15 Mbit/s connection.
Telepresence systems are based on two protocols: SIP (Session Initialisation Protocol) standardised by the IETF (Internet Engineering Task Force) and H.323 standardised by the ITU (International Telecommunication Union).
Telepresence systems from different vendors may follow different implementation arrangements while delivering a similar user experience. They may use different techniques to describe, control and negotiate media for transmission and reception, which can cause interoperability issues where two end-points create the connection. User experience can suffer, for example, if video images captured by cameras are displayed in the wrong order. Even when the telepresence system uses the same protocol SIP, vendors may use proprietary protocol extensions to overcome telepresence related problems.
For these reasons, CLUE (Controlling Multiple Streams for Telepresence) has been created. CLUE is a set of specifications to enable interoperability between telepresence systems by addressing the following issues:
- Description of spatial arrangement of captured video;
- Description of spatial arrangement of captured audio;
- Individual audio streams associated with one or more video captures and individual video captures to be associated with one or more audio captures;
- Interoperability between end-points that have different numbers of devices;
- Interoperability between end-points with different video capture aspect ratios;
- Information that enables rendering of a video image at the actual size of the captured scene;
- Interoperability between telepresence endpoints where displays are of different resolutions;
- Handling different bit rates in the same conference;
- Interoperability between endpoints that send and receive different numbers of media streams;
- Endpoints that support telepresence extensions can establish a session with a SIP endpoint that does not support the telepresence extensions;
- Mechanism for determining whether or not an endpoint or MCU is capable of telepresence extensions;
- Means to enable more than two endpoints to participate in a teleconference;
- Support both transcoding and switching approaches for providing multipoint conferences;
- Mechanisms to allow media from one source endpoint and/or multiple source endpoints to be sent to a remote endpoint at a particular point in time;
- Supporting presentations with different media sources, presentations for which the media streams are visible to all endpoints and multiple simultaneous presentation media streams, including presentation media streams that are spatially related to one another.
Telepresence implementation in IMS networks
IMS uses an IETF defined session control mechanism with the capability to negotiate multiple media streams in one session which is applied to support telepresence in the IMS network environment. IMS based telepresence incorporates CLUE with SIP and SDP to accommodate control of multiple spatial media streams in an IP media telepresence session.
SDP protocol is used in IMS to establish multimedia streams. In a telepresence session, each end-point sends and receives multiple multimedia streams which may be asymmetric, as each point may have different capabilities for media production.
For a telepresence session, a CLUE data channel needs to be established. This channel is used for transportation of bidirectional CLUE messages. This channel must be established before any CLUE protocol message or CLUE-controlled media is sent or received. Once a CLUE data channel is established, CLUE protocol messages can be exchanged between end-points to advertise and configure audio and video components in the telepresence session.
Figure 1 shows an example of message flows in a telepresence session. Any update to the telepresence session, such as a change of media streams, is done by the CLUE data channel where the information intended for streaming and receiving is exchanged with CLUE ADVERTISEMENT and CLUE CONFIGURE messages. Based on this, media streams are changed via SDP offer in SIP re-INVITE message.
Test solution
Telepresence support in an IMS network gives the opportunity for telepresence services using different kinds of devices including mobile systems, presenting a new set of challenges for users who need to test and verify these services. Fortunately, the latest generation of communications test equipment such as the Anritsu MD8475A and MD8475B signal analysers (Figure 2) provide the test environment for such complex mobile services, including those provided over IMS networks.
These instruments offer the capability of testing VoLTE, conference calling, public safety answering point, rich communication services and other IMS features, putting the user in a strong position to verify, test and troubleshoot many features and functions. The MX847570A-060/MX847570B-060 IMS script basic option simulates CSCF, Virtual UE and XCAP with the flexibility for creating customised IMS scripts for testing specific scenarios. A RTP Frame Control option can also offer more flexibility in testing where delay and loss of RTP packets are introduced to the voice or video stream of the call, thereby getting closer to real life scenario simulation. Functionality such as setting different patterns of data in downlink streams and sending periodic silent indicators in downlink again help get closer to real life scenario simulation and extend the range of device testing. Based on the extended functionality of the Anritsu signalling tester, new fields of testing emerge for power consumption testing. Mobile devices can be easily tested to determine how much energy the DUT consumes in different call conditions.
Further opportunity for enhanced testing capabilities is provided by the ability to connect to an external IMS test network where an Anritsu signalling analyser can serve as an access network. With this connection, the user can easily test and verify the interoperability of various telepresence systems with the test device. The testing of device functionality in telepresence system is not only limited to normal or so-called positive testing but abnormal and negative scenarios can also be included. With an IMS test network capable of simulating different signalling messages for telepresence configuration, a wide range of test cases can be run against the device. Typical examples of such test cases are sending invalid messages or ignoring received messages during the setup or reconfiguration of a call which offer additional scenarios for testing the behaviour of the device.
One other possible test example is to measure the quality of a voice in VoLTE call. The concept of measurement in this case is very simple: capture and analyse data packets. Once data packets of a call are captured, they can be decoded and analysed with different matrices to provide useful performance indicators. One of well known solution for voice quality testing is ACQUA from HEAD acoustics which is integrated in the Anritsu signalling tester MD8475A.
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