RTP is a protocol for transporting real-time multimedia data like audio and video over IP networks. It works with RTCP for feedback on data delivery. RTP packet headers include fields for payload type identification, sequence numbering, timestamps, and SSRC for stream source identification. SIP establishes and manages multimedia sessions and calls over IP, allowing participants to negotiate media encodings and manage calls through actions like adding streams, changing encoding, or inviting others. SIP messages are ASCII text sent over UDP or TCP and it requires message acknowledgment.
3. Real Time protocol
• Real Time protocol ( RTP )
• can be used for transporting common formats such as PCM, ACC, and MP3 for
sound and MPEG and H.263 for video
• Packet structure for carrying audio/video data
• RTP provides
• Payload identification
• Packet sequence numbering
• Time stamping
4. Real Time protocol ( RTP )
• RTP packet has UDP segment and media chunk
• RTP is for end systems only, intermediate routes don’t examine it
• RTP in end systems and RTP libraries provides transport layer interface that extends
UDP with
• Port numbers, IP address
• payload type identification
• packet sequence numbering
• time-stamping.
6. RTP headers
• Synchronization source identifier (SSRC). The SSRC field is 32 bits long. It
identifies the source of the RTP stream. Typically, each stream in an RTP
session has a distinct SSRC.
• The SSRC is not the IP address of the sender, but instead is a number that
the source assigns randomly when the new stream is started.
• The probability that two streams get assigned the same SSRC is very small.
Should this happen, the two sources pick a new SSRC value
7. Real-Time Control Protocol (RTCP)
• Used in combination with RTP
• All participants send reports periodically to all others , number of packets
lost/sent, inter arrival jitter. Timestamps at receiver Vs RTP media
timestamps etc.
• Scaling issue is in multicasting. Too many packets by receivers
• RTCP modifies the rate with which participants send traffic into the
multicast tree as a function of the number of participants in the session
8. Session Initiation protocol (SIP)
• Mechanisms for establishing calls over an IP network.
• Allows the caller to notify the callee that it wants to start a call.
• Allows the participants to agree on media encodings.
• Allows participants to end calls.
• Mechanisms for the caller to determine the current IP address of the callee.
• Users do not have a single, fixed IP address because they may be assigned addresses
dynamically (using DHCP) and because they may have multiple IP devices, each with a
different IP address.
9. Session Initiation protocol (SIP)
• Mechanisms for call management
• such as adding new media streams during the call
• changing the encoding during the call
• inviting new participants during the call
• call transfer, call holding.
10.
11. Key Characteristics of SIP
• First, SIP is an out-of-band protocol: The SIP messages are sent and received
in sockets that are different from those used for sending and receiving the
media data.
• Second, the SIP messages themselves are ASCII-readable and resemble
HTTP messages.
• Third, SIP requires all messages to be acknowledged, so it can run over UDP
or TCP
12. Scenarios
• Bob doesn’t have proper codec
• Will send list of available codecs, Alice will choose one and resend invite with that
codec
• Busy
• Gone
• Payment required
• Forbidden etc.
13. SIP
• IP not known to Alice
• Will send request on email id like bob@domain.com
• SIP Proxy will respond with IP address of Bob or voicemail box or a URL saying Bob
is sleeping
• SIP registrar.
• Every SIP user has an associated registrar. Whenever a user launches an SIP application
on a device, the application sends an SIP register message to the registrar, informing the
registrar of its current IP address
