1800 telephone   1900 telephone      1970' telephone              2000 telephone      1990 Cellular        1995 Cellular          2007 Cellular

 

 

 

Early room system         4 way system                  Content+video               LCD system              Desktop VC                          HD VC

 

Standards: SIP vs H.323

There are two major standards within video conferencing industry. The H.323 standard was the first standard to introduce IP video conferencing and has gradually replaced the H.320 standard that was intruduced for the early ISDN and POTS system that entered the market. IP video conferencing has taken over the video conferencing arena. SIP was developed with the introduction of VoIP (voice over IP services) as a simplified alternative to the rivalling H.323 standard. SIP has simplified and resolved some of the issues found within IP video conferencing, like firewall traversal and other challenges introduced with today's advanced security features in IP networks. The h.323 standard is still dominating the Video conferencing market, whil the SIP standard is gradually taking over the SIP market. The SIP standard is also prognosed to take over the video conferencing market. Down the line, H.325 is the next step.

 

 

 

	H.323 Video conferencing	SIP Video Conferencign
Architecture	H.323 covers services, such as capability exchange, conference control, basic signaling, QoS, registration, service discovery, and so on.	SIP is modular because it covers basic call signaling, user location, and registration.  Other features are in other separate orthogonal protocols.
Components	Terminal/Gateway	UA
	Gatekeeper	Servers
Protocols	RAS/Q.931	SIP
	H.245	SDP
Call control
Call Transfer	Included	Included
Call Forwarding	Included	Included
Call Holding	Included	Included
Call Parking/Pickup	Included	Included
Call Waiting	Included	Included
Message Waiting Indication	Included	Not included
Name Identification	Included	Not included
Call Completion on Busy Subscriber	Included	Included
Call Offer	Included	Not included
Call Intrusion	Included	Not included
	H.323 splits them across H.450, RAS, H.245 and Q.931
Advanced functionality
Multicast Signaling	Included, location requests (LRQ) and auto gatekeeper discovery (GRQ).	Included, e.g., through group INVITEs.
Third-party Call Control	Included, through third-party pause and re-routing which is defined within H.323. More sophisticated control is defined by the related H.450.x series of standards.	Included, through SIP as described in separate Internet Drafts.
Conference	Included	Included
Click for Dial	Included	Included
Scalability features
Large Number of Domains	The initial intent of H.323 was for the support of LANs, so it was Not inherently designed for wide area addressing. The concept of a zone was added to accommodate wide area addressing.  Procedures are defined for “user location” across zones for email names.	SIP inherently supports wide area addressing. When multiple servers are involved in setting up a call, SIP uses a loop detection algorithm similar to the one used in BGP, which can be done in a stateless manner, thus avoiding scalability issues. The SIP Registrar and redirect servers were designed to support user location.
Large Number of Calls	H.323 call control is commonlyimplemented in a stateless manner.  A gateway can use messages defined in H.225 to assist the gatekeeper in performing load balancing across gateways.	Call control is commonly implemented in a call stateless manner. SIP supports n to n scaling between UAs and servers. SIP takes less CPU cycles to generate signaling messages; therefore a server could theoretically handle more transactions. SIP has specified a method of load balancing based upon the DNS SRV record translation mechanisms.
Connection State	Stateful or Stateless.	Stateful or Stateless.  A SIP call is independent of the existence of a transport-layer connection, but instead signals call termination explicitly.
Internationalization	Included, H.323 uses Unicode (BMPString within ASN.1) for some textual information (h323-id), but generally has few textual parameters.	Included, SIP uses Unicode (ISO 10646-1), encoded as UTF-8, for all text strings, affording full character set neutrality for names, messages and parameters. SIP provides for the indication of languages and language preferences.
Securityfeatures	Defines security mechanisms and negotiation facilities via H.235, can also use SSL for transport-layer security.	SIP supports caller and callee authentication via HTTP mechanisms. Cryptographically secure authentication and encryption is supported hop-by-hop via SSL/TSL, but SIP could use any transport-layer or HTTP-like security mechanism, such as SSH or S-HTTP. Keys for media encryption are conveyed using SDP. SSL supports symmetric and asymmetric authentication. SIP also defines end-to-end authentication and encryption using either PGP or S/MIME.
Interoperability	The fully backward compatibility in H.323 enables all implementations based on different H.323 versions to be seamlessly integrated. H.323 provides an implementers’ guide, which clarifies the standard and helps towards interoperability among different implementations.	In SIP, a newer version may discard some old features that are not expected to be implemented any more. This approach saves code size and reduces protocol complexity, but loses some compatibility between different versions.  SIP thus far has not provided an implementation agreement.
Commercial services (billing)	Even with H.323's direct call model, the ability to successfully bill for the call is not lost because the endpoint reports to the gatekeeper the beginning and end time of the call via the RAS protocol.	If the SIP proxy wants to collect billing information, it has no choice but to stay in the call signaling path for the entire duration of the call so that it can detect when the call completes. Even then, the statistics are skewed because the call signaling may have been delayed.
Codecs	H.323 supports just about any codec, standardized or proprietary, not just ITU-T codecs. There have been codepoints for MPEG and GSM, which are not ITU-T codecs, in H.323 for a long time; many vendors support proprietary codecs through ASN.1 NonStandardParameters, which is equivalent to SIP's "privately-named codec by mutual agreement"; and any codec can be signaled via the GenericCapability feature that was added in H.323v3. Payload types can be specified statically or dynamically. Examples: H.263, H.264, G.711,G.722,G.723, G.728, G.729, AAC	SIP supports any the IANA-registered codecs (as a legacy feature) or other codec whose name is mutually agreed upon. Payload types can be specified statically or dynamically. Examples: H.263, H.264, G.711,G.722,G.723, G.728, G.729, AAC
Call Forking	H.323 gatekeeper can control the call signaling and may fork the call to any number of devices simultaneously.	SIP proxies can control the call signaling and may fork the call to any number of devices simultaneously.
Transport protocol	Reliable or unreliable, meaning TCP or UDP. Most H.323 entities use a reliable transport for signaling.	Reliable or unreliable, meaning TCP or UDP. Most SIP entities use an unreliable transport for signaling.
Message Encoding	H.323 encodes messages in a compact binary format that is suitable for narrowband and broadband connections.	SIP messages are encoded in ASCII text format, suitable for humans to read.
Addressing	Flexible addressing mechanisms, including URLs and E.164 numbers.	SIP only understands URL-style addresses.
PSTN Interworking	H.323 borrows implementations from traditional PSTN protocols, e.g., Q.931, and is therefore well suited for PSTN integration. However, H.323 does not employ the PSTN's circuit-switched technology--like SIP, H.323 is completely packet-switched.	SIP has no commonality with the PSTN and such signaling must be "shoe-horned" into SIP. SIP has no architecture that describes the decomposition of the gateway into the Media Gateway Controller and the Media Gateways.
Loop Detection	Included, routing gatekeepers can detect loops by looking at the CallIdentifier and destinationAddress fields in call-processing messages. If the combination of these matches an existing call, it is a loop.	Included, the SIP message Via header facilitates this. However, there has been talk about deprecating Via as a means of loop detection due to its complexity. Instead, the Max-Forwards header seems to be the prefered method of limiting hops and therefore loops.
Minimum Ports for VoIP Call	5 (Call signaling, 2 RTP, and 2 RTCP.)	5 (Call signaling, 2 RTP, and 2 RTCP.)
Video and Data Conferencing	H.323 fully supports video and data conferencing. Procedures are in place to provide control for the conference as well as lip synchronization of audio and video streams.	SIP has no support for data conferencing protocols like T.120, although there are emerging standards for this .
Microtronix Test System Available	Included	Included

 

Pixavi's implementation

Pixavi has implemented both SIP and H.323 protocols into our products, in order to provide the best possible interoperability towards the video conferencing products on the market.

 

Video Conferencing Codecs

ITU has standardized on several codecs for use within Video Conferencing. The latest video codec: H.264 and the latest audio codecs: AAC, provides exceptional quality and improves the user experience significantly.

 

 

signaling standards	Video codecs	Voice codecs
SIP H.323 ISMA streaming	H.261 H.263 H.264	G.711 G.722 G.722.1 G.728 AAC-LD AAC-LC

 

Pixavi codecs

Pixavi has implemented all the above signaling standards and codecs in our communication products, for maximum performance, quality and interoperability in our wireless video conferencing products.

 

 

Other Video conferencing signaling implementations:

SCCP

Skinny Client Control Protocol (SCCP) is a Cisco proprietary protocol used between Cisco Call Manager and Cisco VOIP phones. It is also supported by some other vendors. For VOIP solutions, the end station of a LAN or IP- based PBX must be simple to use, familiar and relatively cheap. SCCP defines a simple and easy to use architecture, while the H.323 recommendations are quite an expensive system. An H.323 proxy can be used to communicate with the Skinny Client using the SCCP. In such a case the telephone is a skinny client over IP, in the context of H.323. A proxy is used for the H.225 and H.245 signaling. With the SCCP architecture, the vast majority of the H.323 processing power resides in an H.323 proxy known as the Cisco Call Manager. The end stations (telephones) run what is called the Skinny Client, which consumes less processing overhead. The Client communicates with the Call Manager using connection-oriented (TCP/IP-based) communication to establish a call with another H.323-compliant end station. Once the Call Manager has established the call, the two H.323 end stations use connectionless (UDP/IP-based) communication for audio transmissions. Costs and overhead are thus reduced by confining the complexities of H.323 call setup to the Call Manager, and using the Skinny protocol for the actual audio communication into and out of the end stations.

 

H.324m

The 3G-324M protocol operates over an established circuit switched connection between two communicating peers. 3G-324M is an umbrella specification to enable conversational multimedia communication over Circuit Switched (CS) networks and has been adopted by the 3GPP. 3G-324M is based on the ITU-T H.324 specification for multimedia conferencing over Circuit switched networks. 3G-324M is agnostic to the actual Circuit switched network that uses it. It can run as easily over UMTS as well as TD-SCDMA networks. 3G-324M is a proven solution for conversational multimedia based services that packet-based wireless networks cannot deliver because of inherent overhead, BER sensitivity, and variant routing delays. 3G-324M operating over a circuit switched channel between two communication peers guarantees the fixed-delay quality of service for multimedia communications. Combining Circuit switched 3G-324M services with packet-based SIP services such as presence can leverage the strength of both networks to enable new types of differentiated and innovative mobile 3G services.

 

AMS

Advanced Multimedia System (AMS) is a new multimedia system driven as a project by the ITU and named Q12/16. Presently, the ITU is working on an architecture, studying usage scenarios, and doing preliminary work on protocol definitions. It has been decided that the system will use XML for signaling.

 

The AMS project will give us the generation multimedia terminal and system architectures able to support emerging, multimedia applications that is totally different from call-based communication platforms. These applications include tightly integrated and convergenced media applications involving multiple personal and public devices, enterprise systems and network services for use of communications, collaboration and entertainment. Specifications arising from this project will enable the development of the terminals and systems, and also inter-communication between systems so applications involving multiple devices and mobile systems can be supported.

 

The goal of the AMS project is to create a new multimedia terminal and systems architecture that supports distributed and media rich collaboration applications over public IP network (Internet). Current interactive multimedia protocols (H.323 and SIP) adds media to call-based communication establishment protocols enabling multimedia telephony.

 

In contrast, AMS outlines functionality which lets a user have many AMS-enabled devices including portable wireless, home entertainment and computer-based devices and is offered several applications and services that are either peer-to-peer or network-provided. The users coordinates activities across all of these environments using the modes that best fit their personal and business situation and needs or desires.

 

AMS is viewed as the successor system to H.323 and SIP systems, which are now more than a decade old. AMS is still being defined, so it is very much in its early stages. It is expected to solidify over the next two to three years, with an initial release targeted for 2010.

 

Endpoint revolution

Traditional video conferencing endpoints are no longer the only endpoints able to conduct video conferences. Computers, mobile phones, PDA's and other devices are now equiped with video conferencing software that can cover some of the applications in today's video conferencing market. The future holds an increased number of innovative devices and software applications for video communication and collaboration.

 

 

Mobile phones        Video conferencing           Windows computer             Alternate OS devices           Set top Box

 

 

 

Mobile wireless video conferencing

Mobile phones are low cost devices that are optimized for download experiences and audio communication. Bound by the limitations of slow networks, subscription costs and limited processing power, the user can not expect the quality offered by dedicated video conferencing products.

 

Video Quality and mobile video conferencing

Mobile phone networks are not optimal for conducting video conferencing other than low quality face to face conferencing. Cellular networks are optimized for download speed and not upload speeds. For high quality video conferencing, other wireless standards are preferable.

 

Combining wearable technology, wireless technology, high end optics, encoding power, collaboration technology and imaging sensors, Pixavi has created innovative video conferencing products since 1999. The latest product, dubbed the Xcaster, offers HD quality video over wireless networks.

 

 

 

Pixavi Xcaster ST5000 wireless High Definition video conferencing.

 

 

 

From Office to Field and from Field to Office without travel

Today's major threats to world piece are lack of energy resources, terrorism and climate changes. All of these threats will constrict how people travel in the future. video conferencing, collboration and telepresence are technologies that can help people conduct their private and professional tasks without sacrificing quality, safety and end results.

 

Challenges lies within cultural and interactional traditions. Convincing people to use Video Conferencing when a travel is their natural choice is not easy. Video communication must become a natural communication form before it can start having real impact on the climate and energy consumption on a global basis.

 

 

Video conferencing room

 

Telepresence

Telepresence is in reality high quality video conferencing that uses several video streams to recreate the surroundings of two or more locations. The result is a unique user experience. The future holds telepresence for everyone..

 

Mobile Telepresence

Pixavi has introduced a mobile telepresence device that, unlike other wireless IP video products, provides exceptional audio, video and still image quality, functionality and performance.

 

 

Firewall traversal technologies

 

Port tunneling

When streaming video using the ISMA standard one can tunnel all the IP packets through Port 80. This is the same port that receives HTTP traffic, indicating that most firewalls have opened up for traffic on this port. Pixavi implements streaming video over port 80 for improving firewall traversal when streaming video to Windows Media player, Quicktime or VLC

 

STUN

STUN (Simple Traversal of UDP through NATs (Network Address Translation)) is a protocol for assisting devices behind a NAT firewall or router with their packet routing.

• STUN enables a device to find out its public IP address and the type of NAT service its sitting behind.

• STUN operates on TCP and UDP port 3478.

• STUN is not widely supported by VOIP devices yet, but this is changing rapidly

• STUN may use DNS SRV records to find STUN servers attached to a domain. The service name is _stun._udp or _stun._tcp

 

 

Definitions (from the RFC)

• STUN Client: A STUN client (also just referred to as a client) is an entity that generates STUN requests. A STUN client can execute on an end system, such as a user's PC, or can run in a network element, such as a conferencing server.

• STUN Server: A STUN Server (also just referred to as a server) is an entity that receives STUN requests, and sends STUN responses. STUN servers are generally attached to the public Internet.

Various types of NAT (still according to the RFC)

• Full Cone: A full cone NAT is one where all requests from the same internal IP address and port are mapped to the same external IP address and port. Furthermore, any external host can send a packet to the internal host, by sending a packet to the mapped external address.

• Restricted Cone: A restricted cone NAT is one where all requests from the same internal IP address and port are mapped to the same external IP address and port. Unlike a full cone NAT, an external host (with IP address X) can send a packet to the internal host only if the internal host had previously sent a packet to IP address X.

• Port Restricted Cone: A port restricted cone NAT is like a restricted cone NAT, but the restriction includes port numbers. Specifically, an external host can send a packet, with source IP address X and source port P, to the internal host only if the internal host had previously sent a packet to IP address X and port P.

• Symmetric: A symmetric NAT is one where all requests from the same internal IP address and port, to a specific destination IP address and port, are mapped to the same external IP address and port. If the same host sends a packet with the same source address and port, but to a different destination, a different mapping is used. Furthermore, only the external host that receives a packet can send a UDP packet back to the internal host.

 

 

H.460 firewall traversal

H.460.18, H.460.18 and H.460.19 are standards from ITU that enable H.323 devices to exchange signaling and media across boundary imposed by NAT and firewalls, with no additional equipment on the customer premise.