Voice over Internet Protocol - VoIP
Trillium Voice over Internet Protocol (VoIP) software source code solutions enable the convergence of the IP and PSTN networks.
VoIP refers to a new class of applications that merge Internet capabilities with PSTN functions. While VoIP focuses primarily on voice services, it can also be used to carry other voice-band and multimedia applications such as fax, video and data. Convergence of the Internet and PSTN provides more effective network usage, substantial cost savings and many new revenue opportunities. Trillium VoIP solutions consist of H.323, MGCP and SIP.
Trillium-Powered VoIP Network Elements
- Cable modem
- Gatekeeper (GK)
- Integrated Access Device (IAD)
- IP Phones/Appliances
- Media Gateway (MG)
- Media Gateway Controller (MGC)
- Media Server
- Multipoint Conference Unit (MCU)
- Multi-service Switch
- Session Border Controller (SBC)
- Softswitch
- Signaling Gateway (SG)
Trillium VoIP Protocols
- H.248 / MEGACO
- H.323
- MGCP - Media Gateway Control Protocol
- RTP/RTCP/SRTP - Real-Time Protocol / Real-Time Control Protocol / Secure Real-Time Protocol
- TUCL - TCP/UDP Convergence Layer
H.323
H.323 is a standard from the International Telecommunications Union (ITU-T) that specifies the components, protocols and procedures for multimedia communication over packet-based networks such as IP-based Local Area Networks (LANs) and Wide Area Networks (WANs), including the Internet. Trillium H.323 source code solutions enable the implementation of VoIP devices.
H.323 Stack Diagram
H.323 Stack Components
H.245 Control Signaling:
- Control signaling between endpoints (client, Gateway) or between endpoints and Gatekeeper/MCU to determine the capabilities of the communicating endpoints
- Opening and closing logical channels for media streams
- Conference control commands
H.225 RAS:
- Registration, Admission, and Status (RAS) control signaling between an endpoint (client, Gateway) and Gatekeeper
RTP:
- Real-time Transport Protocol (RTP) for carrying real-time audio/video data
RTCP:
- Real-time Transport Control Protocol (RTCP) for for providing feedback on the transmission and reception quality of data carried by the RTP
Audio/Video Codecs:
- Encoding/Decoding audio/video signals to and from the H.323 Terminal
H.450 Supplementary Services:
- Adds basic call management features to a Terminal such as Call Transfer, Caller ID, and Call Forwarding (RTCP) for providing feedback on the transmission and reception quality of data carried by the RTP
H.323 Solutions
Trillium H.323 Control supports:
- H.323v3, H.225v3 Call Signaling, H.245v5 Control Signaling, H.225v3 RAS
- All H.450 Supplementary Services
- Annex E-based call signaling over UDP
- Annex G-based Gatekeeper-to-Gatekeeper (inter-domain) communication
- Flexible RAS API that allows minimal or full control for device implementers
- Fast connection procedures
- Tunneling of H.245 massages in the H.255.0 Call Signaling PDU's
- Interfaces for developing and deploying Gateway, Gatekeeper, multipoint controller, and Terminal services
- Static and round-robin distribution of incoming calls
- Transparent Gatekeeper-routed call mode
- API's for H.235 security protocol services
- Coexistence of multiple H.323 entities such as Gatekeeper and Gateway, using the same instance of the code
- Integrated ASN.1 Encoder/Decoder-PER library
Trillium ASN.1 Encoder/Decoder-PER supports:
- Basic PER aligned variant encoding of ASN.1 data types to generate the transfer syntax, as specified in ITU-T Recommendation X.691
TCP/UDP Convergence Layer (TUCL):
- TUCL portable software product is a generic protocol software layer that can be used as a transport layer with the Trillium H.323 stack
Media Gateway Control Protocol (MGCP)
Trillium Gateway Control Protocol software code solutions include Media Gateway Control Protocol (MGCP) and H.248/MEGACO - products that accelerate the convergence of the Internet and PSTN.
MGCP is an IETF informational draft that has gained wide acceptance in the VoIP market segment. MGCP has also been adopted by the CableLabs PacketCable initiative. An enhanced version of MGCP, known as the H.248/MEGACO protocol, has evolved into a new joint ITU-IETF standard. H.248/MEGACO specifies a device control protocol with support for VoIP, multimedia and conferencing traffic capability.
The Gateway Control Protocol (GCP) is an essential technology for VoIP Gateway solutions. The decomposed Gateway model safeguards investments in legacy telephone networks while enabling the migration to a distributed and flexible IP network architecture.
The GCP solution represents a simplified messaging path between the Media Gateway Controllers (MGCs), also referred to as Call Agents (CAs), and Media Gateways (MGs). GCP functions in a call control architecture consisting of signaling gateways that are responsible for PSTN control and intelligent MGCs controlling reduced intelligence MGs.
MGCP and H.248/MEGACO allow for the "decomposition" of telephony gateways by separating the signaling and service capability (provided by the MGC), SS7 termination (provided by the SG) and the media transmission capability (provided by the MG). This model of a decomposed Gateway allows for the convergence of legacy telephony networks with an IP-based, next-generation network infrastructure.
MG Stack Diagram
MGC Stack Diagram
MGCP Solutions
Trillium GCP conforms to:
- IETF-established RFC 2705 MGCP standard
- CableLabs PacketCable Network Call-based Signaling Protocol Specification (NCS)
- Joint ITU-IETF RFC 3015 H.248/MEGACO standard
Common MGC and MG features:
- APIs to build MGCs and MGs
- Encode/Decode library engine for transmission and reception of all messages using text encode/decode
- Encodes/Decodes SDP descriptions in messages
- Manages the MGCP transactions over the UDP
- Supports H.248/MEGACO over the TCP
- Supports configured and discovered MGs
- Supports virtual and physical MGs in one instance
- Provides failover/service change support
MG features:
- Support for all defined packages
- Supports interaction with the configured MGC
- Supports protocol operation on the default port (protocol-specific) or on any other configured port
- Provides failover support
- RTP/RTCP for the media layer to transfer media
MGC features:
- Supports management of multiple MGs within a single instance
- Supports a distributed call control application for managing multiple MGs
- Supports communication on the default port or any other user-selected port
- Support for managing the transaction load on the MGC
Trillium TCP/UDP Convergence Layer (TUCL) provides:
- Common transport functions for applications operating over the TCP/UDP/IP stacks
- Provides transparent mapping to the underlying socket interface
- Direct mapping to the socket interface for operating systems such as Linux, Solaris, Windows, VxWorks, and pSOS
Session Initiation Protocol (SIP)
Trillium Session Initiation Protocol (SIP) software source code solutions are comprehensive and support both SIP and SIP-BCP-T Protocols.
SIP is an IETF standard that specifies the basic and supplementary services to create, modify and delete multimedia sessions or calls. These multimedia sessions include, but are not limited to, multimedia conferences, distance learning and Internet telephony. The SIP-BCP-T is an IETF-defined extension of SIP that enables inter-MGC communication.
SIP is a lightweight, transport-independent, text-based protocol used as a signaling protocol for Internet conferencing and telephony. SIP has only six different types of methods, which reduces the level of complexity for the user. It can be used with any datagram or stream protocol such as UDP, TCP, or ATM, thereby providing flexibility of use. Lastly, because SIP is text-based it is considered a low overhead protocol. All of these factors allow for seamless call routing -- based on the currently specified call flow instructions -- and contribute to making SIP highly scalable.
SIP Stack Diagram
SIP Network Architecture
SIP Stack Components
User Agent Client (UAC)
- Supports the caller application that initiates and sends the SIP requests
User Agent Server (UAS)
Receives and responds to SIP the requests on behalf of the clients
- Accepts, redirects, or refuses the calls
Proxy Server
- Receives requests from clients and forwards them to next-hop servers
- Contains the UAC and UAS
Redirect Server
- Accepts the SIP requests, maps the address into zero or more new addresses, and returns those addresses to the client
- Does not initiate SIP requests or accept calls
Registrar/Location Server
- Provides information about a caller's possible locations to redirect and proxy servers
- Capable of being co-located with a SIP server
SIP Solutions
Trillium SIP complies with the core specification, RFC 3261, and SDP RFC 4556 and supports:
- Various SIP entities such as User Agent-Client and Server, Proxy Server, Redirect Server, Registrar
- Extensive APIs to build various SIP entities
- Fast and robust ABNF message parsing
- Any SIP message compliant with the SIP BNF
- TCP and UDP transport
- SIP extension mechanisms such as:
- MIME support for ISUP encapsulation used in SIP-BCP-T
- INFO method for the PSTN gateways
- 183 session-progress method
- SDP extension for Quality of Service (QoS)
- Reliable, provisional responses with the UDP is used as a transport
Trillium TCP/UDP Convergence Layer (TUCL) is a generic protocol software layer that can be used as a transport layer with the SIP layer stack. It provides transparency to the underlying TCP/UDP/IP stack interfaces and enables portable implementation of the TCP/UDP/IP applications.
- RTP data transport per RFC 1889
- Services for real-time data including payload identification, sequence numbering, time-stamping, and delivery monitoring
- Interfaces to create, modify, or terminate media sessions
- UDP and ATM transport mode for RTF data
- Sending individual RTCP packets by a user
