E0262 - Multimedia Communications Anandi Giridharan Electrical Communication Engineering, Indian Institute of Science, Bangalore – 560012, India Multimedia.

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E Multimedia Communications Anandi Giridharan Electrical Communication Engineering, Indian Institute of Science, Bangalore – , India Multimedia Communications Voice over IP

E Multimedia Communications Voice over IP (Real time protocols) Internet Telephony called Voice over IP most important interactive Multimedia application. The term Internet telephony specifically refers to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN). Example: Chatting over Internet (PC to PC). Voice chat.. Decent conversion, Data rate of 8000 bytes per second generated during each talk spurt (64 Kbps Bandwidth).

E Multimedia Communications Packets for voice get generated only during the talk spurts, Every 20msec sender collects the data into chunks 160 bytes/chunk appxly for 8kbytes/sec. Application-layer header is added to each chunk. The data chunk and the header is encapsulated into a UDP packet and transmitted. UDP is used to prevent un-predicted packet loss etc.

E Multimedia Communications Packet Loss Analysis for quality loss Two main reason for packet loss 1. Normal packets loss,  IP packets are lost are not delivered at the destination. 2. Loss due to excessive delay An IP packet arrives, but too late to be played better to drop such packets.  Delays < 150 msec are normally not detected.  Delays > 400 msec can be annoying.(gaps and breaks in voice)  Depending on encoding technique, packet loss rate of up to 20% can be tolerated.

E Multimedia Communications How Jitters are handled 1. By Using sequence number with each packet. We can avoid out-of-order playback. 2. By using Time-stamps in the packet header. Delaying the playout. The playout of packets are delayed so that Protocols Used a. Session Initiation Protocol (SIP) b. ITU standard H.323 Most of the packets arrive before time out.

E Multimedia Communications Session Initiation Protocol(SIP) SIP is an application layer protocol. Used to establish, manage and terminate multimedia sessions. Two party, multi-casting, multi-party (different sessions can be supported for various applications) SIP can run on either TCP or UDP based on QoS.

E Multimedia Communications Sender /Receiver Addressing Instructs a SIP client to make a (usually UDP) connection to voip-provider.example.net:5060 (which may be a gateway) and ask to be connected to the destination user at The gateway may require the user REGISTER using SIP before placing this call. SIP specifies Address: IP address, address, telephone number to identify sender and Receiver. Specified in SIP standard address format.

E Multimedia Communications Basic messages sent in the SIP environment INVITE – connection establishing request ACK – acknowledgement of INVITE by the final message receiver BYE – connection termination CANCEL – termination of non-established connection REGISTER – UA registration in SIP proxy OPTIONS – inquiry of server options

E Multimedia Communications Simple SIP Session Three Steps- Establishing a session Uses a 3-way handshake protocol. Communication Caller and callee uses two temporary ports for the purpose. Terminating the session Either party can initiate this.

E Multimedia Communications Exchange of voice pkts Caller Callee Invite Respond Bye

E Multimedia Communications The H.323 Standard

E Multimedia Communications A standard that allows telephones on the public network to talk to computers on the Internet. There will be Gateway and Gatekeeper in your network That will take care of synchronization Uses a gateway: Connects the telephone network to the Internet. Translates messages from one protocol stack to another. (voice pkts are generated by one kind of protocol) while sending out some translation is required while sending outside.

E Multimedia Communications The H.323 voice/video over IP environment

E Multimedia Communications Internet Gateway Telephone Nw

E Multimedia Communications Various Protocols Used H.323 uses a number of protocols: G.71 or G723.1 Used for compression. H.245 Allows parties to negotiate the compression method. Q.931 For establishment and termination of connections. H.225 Used for registration with the gatekeeper.