 Figure 6.3 illustrates the H.323 architecture, which incorporates a family of standards including H225, H245 and H450. As an international standard for conferencing over packet networks H.323: acts as a single standard to permit Internet telephony products to inter-operate; also serves as base for standard interoperability between ISDN- and telephony- based conferencing systems; and has the flexibility to support different HW/SW and network capabilities. The logical channels in H.323 get multiplexed at the destination port transport address level. The transport address results from the combination of a network address and a port identifying a transport level endpoint, e.g., an IP address and a UDP port. Packets having different payload types go to different transport address, thereby eliminating The UMTS Network and Radio Access Technology usage of separate multiplexing/demultiplexing layer in H.225.0. The H.225 standard uses RTP/RTCP2 for media stream packetization and synchronization for supporting LANs. This usage depends on the usage of UDP/TCP/IP. BER control takes place at lower layers; thus, incorrect packets do not reach the H.225 level. When both audio and video media act in a conference, they transmit using separate RTP sessions, and RTCP packets get transmitted for each medium using two different UDP port pairs and/or multicast addresses. Thus, it does not exist direct coupling at the RTP level between audio and video sessions, and synchronised playback of a source's audio and video takes place using timing information carried in the RTCP packets for both sessions. Point-to-point H.323 conference occurs with two TCP connections between the two terminals, i.e. one for call set-up connection and one for conference control and feature exchange. The first connection carries the call set-up messages defined in H.225.0, i.e. the Q.931 channel. After a 1st st TCP connection on a dynamic port, the calling parties establish the second TCP connection to the given port, where the 2nd connection carries the confe