Streaming media expert
Dr. Lavian offers expert consulting services in streaming media, audio and video conferencing, SIP, RTP, telecommunications, and unified communications. Dr. Lavian’s academic background and practical industry experience provide him with a profound depth and breadth of knowledge regarding network communications, multimedia, streaming, and audio/video conferencing. Moreover, He has experience in different aspects of the relationships between Internet Protocols (TCP/IP), streaming applications, protocols, and services, and their underlying communications service architectures, designs, and implementations.
Dr. Lavian’s extensive experience with streaming protocols, audio/video conferencing, and streaming technologies, primarily implemented through the TCP/IP protocol suite, gives him a keen understanding of the logic behind streaming and its component application, transport, network, data link, and physical layers.
Streaming Media Technologies
Dr. Lavian is intimately familiar with the Internet protocol suite (TCP/IP) and is proficient in networking and communications technologies, including:
- Multimedia, audio, and video streaming.
- Video/audio conferencing, streaming media, and music streaming over TCP/IP protocol stack using standard Internet protocols such as TCP, UDP, IP, HTTP, HTML, and Web services.
- Integration of streaming media with SIP, Internet protocols (TCP/IP), and streaming systems as the backbone of unified communicants.
- Integration of such protocols and services with short message technologies. Including text messaging, SMS, MMS, Instant Messaging (IM), and chat applications.
- Implementing such technologies over handsets, mobile devices, smartphones, cellular networks, wireless technologies, architectures, standards, systems, and backend infrastructure.
- Streaming protocols, SIP, RTP, H.323, MGCP, and codecs.
- Bandwidth, Quality of Service (QoS) related to conferencing and media streaming.
The H.323 protocol is an ITU standard for audio/video conferencing that creates a virtual meeting space using traditional desktop computers, laptops, and wireless mobile devices. Therefore, H.323 can scale to support HD video and HD audio with advanced network components designed to provide lower latency, higher quality, and less jitter for a more unified user experience.
This protocol suite utilizes for audio and video calls. H.323 is one of the most widely utilized protocols for real-time group communications. Above all, the underlying architecture consists of a call processing architecture that contains management entities, H.245 gatekeepers, gateways, media gateways, and endpoints.
Similarly, To facilitate making calls with multimedia capabilities over a packet-based network that does not provide Quality of Service (QoS) guarantees. H.323 consists of an umbrella set of standard recommendations.
Session Initiation Protocol (SIP) was primarily developed by the Internet Engineering Task Force (IETF) and is an alternative to ITU Recommendation H.323. It is lighter and utilized for more general purposes. It is a protocol based on HTTP. Above all, SIP uses to control Internet multimedia conferences, Internet telephone calls, and multimedia distribution. The SIP signals and manages interactive communication sessions such as voice, video, chat, and instant messaging.
SIP invitations are the foundation of every SIP phone call. The SIP invite request is the message sent by the party initiating the call, inviting the recipient to a session. SIP headers included in this SIP invite request provide information about the message. Therefore, The SIP invite creates sessions and carries session descriptions. That allows both endpoints to agree on a set of compatible media types. The advantage is SIP is not restricted to a particular media type.
SIP server is also known as a SIP Proxy. The SIP server deals with all of the management of SIP calls in a network and is responsible for taking requests from the user agents to place and terminate calls. It enables user mobility as it allows requests to redirect (proxied) to the user’s current location (as opposed to PSTN phones “locked” to a specific physical location.) Users register their current location with their SIP server.
SIP authentication is a stateless challenge-based mechanism that ensures a user’s identity. The recipient can ask an authentication challenge to force the SIP invite sender to prove his identity before the message process and response. SIP supports end-to-end as well as hop-by-hop authentication and encryption using S/MIME.
Endpoints of a SIP session communicate using unicast, multicast, or a combination of both. Furthermore, SIP is independent of the lower-layer transport protocol, which allows it to take advantage of various transport protocols.
SIP messages comprise a text-based protocol with syntax like HTTP. There are two types of SIP messages: requests and responses. However, The first line of a request contains a method to define the nature of the request. And a Request-URI indicating where the response should be sent.
The RTP standard defines a pair of protocols: Real-Time Transport Protocol (RTP) and RTP Control Protocol (RTCP). RTP utilizes to exchange multimedia data. At the same time, RTCP is the control protocol used to periodically obtain feedback control information regarding the quality of transmission associated with data flows. The RTP and RTCP are UDP-based protocols, meaning no session is maintained, as opposed to TCP. RTP provides end-to-end network transport for applications transmitting real-time data, such as audio and video, over multicast or unicast network services. In conclusion, RTCP is based on the periodic transmission of control packets to all participants in the session.