DiffServ, Multicast, RTP, SCTP, SIP, H.323 & Watermarking
DiffServ Expedited and Assured Forwarding
Two PHBs (Per-Hop Behaviors) are defined for DiffServ:
- Expedited Forwarding PHB: Applied to real-time traffic and related to the guaranteed service transfer capability. It specifies that the departure rate of a class of traffic from a router must equal or exceed a configured rate.
- Assured Forwarding PHB: Applied to elastic traffic and related to the controlled load service transfer capability. It divides traffic into four classes, each guaranteed a minimum amount of bandwidth and buffering.
IP Multicast vs. CDN
There are major differences between IP multicast and Application-Level Multicast (ALM) as deployed in CDNs:
- Multicast Group Identifier:
- IP Multicast: A class-D address identifies a multicast group.
- CDN: A URL or other application-related key identifies a multicast group.
- Multicast Group Members:
- IP Multicast: Users explicitly subscribe to certain multicast content via the IGMP protocol.
- CDN: Requests are redirected to the user’s local CDN server.
- Network Topology:
- IP Multicast: Routers exactly reflect the physical network topology.
- CDN: Servers form a logical overlay network on top of the physical network infrastructure.
- Multicast Routing:
- IP Multicast: Routing relies on the underlying unicast routing protocols and employs simple metrics (e.g., number-of-hops or delay).
- CDN: Routing is much simpler (e.g., minimum-delay-path spanning tree, widest-path spanning tree, etc.).
Interdomain Multicast Routing Protocols
When group members are spread across different domains (AS), an interdomain multicast routing protocol is needed. Examples include:
- Multicast Border Gateway Protocol (MBGP): An extension of BGP and a shared-group multicast routing protocol.
- Multicast Source Discovery Protocol (MSDP)
- Border Gateway Multicast Protocol (BGMP)
RTP and RTCP: Real-Time Interactive Protocols
Fulfillment of requirements with RTP and RTCP:
- Sender-Receiver Negotiation: Cannot be satisfied directly; another protocol (SIP) must be considered.
- Creation of Packet Stream: Each data chunk is encapsulated in an RTP packet with a sequence number.
- Source Synchronization: Satisfied by an identifier and relative timestamp in the RTP packet, and the absolute timestamp in the RTCP packet.
- Error Control: Using the sequence number in the RTP packet, the application can regenerate lost packets using FEC methods.
- Congestion Control: Achieved by feedback from the receiver using RTCP receiver report packets, allowing the sender to adjust compression algorithms.
- Jitter Removal: Achieved by the timestamp and sequence number in each RTP packet to buffer playback.
- Sender Identification: Achieved by the CNAME included in RTCP packets.
SCTP Features and Services
Stream Control Transmission Protocol (SCTP) features and services include:
- Message-Oriented Protocol: SCTP transports a sequence of messages, rather than a stream of bytes like TCP. Similar to UDP, SCTP sends a message in one operation, passed to the receiving application process in one operation.
- Connection-Oriented Service: Like TCP, SCTP is connection-oriented, using an association negotiated before sending data.
- Message Ordering: Optional; applications can process messages in the order of receipt instead of the order of sending, allowing urgent data processing.
- Full-Duplex Communication: Like TCP, data can flow in both directions simultaneously.
- Reliable Service: Like TCP, it’s a reliable transport protocol using an acknowledgment mechanism.
- Path Selection and Monitoring: SCTP selects and monitors a primary data transmission path and tests connectivity.
- Security: Improves security against DoS attacks, such as SYN attacks.
SIP Services
SIP provides the following services:
- Call Establishment:
- Finds the location of users (their IP addresses).
- Determines if users are able or willing to participate in the conference call.
- Determines users’ capabilities in terms of media and encoding types.
- Session Setup: Establishes session setup by defining parameters (ports).
- Session Management: Provides call holding, call forwarding, accepting new participants, and changing session parameters.
H.323 Protocols
H.323 is a complete set of protocols for establishing and maintaining voice (or video) communication, unlike SIP, which is only a signaling protocol. Key components include:
- G.711 or G.723.1: Compression algorithms.
- H.245: Allows parties to negotiate the compression method.
- Q.931: Establishes and terminates connections.
- H.225: Registration with the gatekeeper.
- RTP and RTCP: H.323 mandates the use of these protocols.
SIP vs. H.323
Key differences between SIP and H.323:
- H.323 defines a complete set of protocols for real-time interactive multimedia (signaling, registration, admission control, transport, and compression). SIP is primarily for initialization, management, and termination of sessions.
- H.323 was defined by ITU-T (telephony), while SIP was proposed by IETF (Internet).
- H.323 is a complex standard, while SIP is simpler and easier to implement.
Digital Watermarking
A watermark is created and inserted into the original digital content to produce watermarked content.
Watermark Embedding Techniques
Two main watermark embedding techniques exist:
- Coefficient-Based Approaches: Embedding is performed by directly modifying pixel values.
- System-Based Techniques: Embedding is performed by slightly changing an existing processing system.
Coefficient-Based Watermarking
Embedding is performed by directly modifying pixel values.
- Simplest method: The watermark is added to the original media similarly to additive noise.
- Other methods insert the watermark after the media has been transformed.
Watermark Insertion Methods
- Spatial Domain: Watermark is inserted by operating on the pixels of the image or video.
- Transformation Domain: Watermark is inserted after a transformation, including techniques like:
- Discrete Fourier Transform (DFT)
- Discrete Cosine Transform (DCT)
- Wavelet Transform (WT)