IP Telephony Endpoints

Major IP Telephony Endpoints (my quick notes)

» Analog Gateways
» Cisco Unified IP Phones
» Software-Based Endpoints
» Wireless Endpoints
» Cisco IP Conference Station
» Video Endpoints

Reference:

» IP Telephony Endpoints

Filed under: » Gatekeeper, » Gateway, » Voice

Review Notes: About H.323 – Specifications

H.323 is a standard for communication protocols from the International Telecommunications Union Telecommunication Standardization Sector (ITU-T); Version 4 is the current version. H.323 was created to provide multimedia communication across a packet network. The protocol can handle video and data, in addition to audio.

Gateways that use H.323 do not depend on a call agent, as with Media Gateway Control Protocol (MGCP). H.323 is the default gateway protocol on Cisco routers.

Because gateways function as H.323 endpoints, they provide admission control, address lookup and translation, and accounting services.

In an environment in which both gatekeepers and gateways are used, only gateways are configured to send VoIP.

H.323 Specifications

• H.225: Handles call setup and teardown between H.323 devices on a packet-based network, terminal to gatekeeper signaling using Registration, Admission, and Status Protocol (RAS), and call signaling. H.323 can use ISDN Q.931 signals, formatted as H.225 messages, to interoperate with legacy voice networks.

• H.235: Specifies security for messages between the gateway and gatekeeper.

• H.245: Controls the traffic flow, performs DTMF Relay, limits media transmission rates, negotiates capability, and controls opening and closing channels for media streams. Uses TCP.

• H.261/ H.263: Specify video conferencing standards.

• H.450: Controls supplementary services between H.323 entities. Examples of supplementary services include call waiting, hold, transfer, park, and pickup.

• T.120: Used for real-time multipoint data transfer during videoconferences. Allows application sharing, whiteboarding, and file transfer. Uses TCP.

• H.320: Defines the standard for video conferencing over ISDN networks. H.320 uses H.221 frames for media. It requires a gateway to interwork with H.323 conferencing over IP because H.221 frames must be translated into RTP packets, and vice versa.

Filed under: » CCVP, » Gateway, » H.323, » Review Notes, » Voice

Review Notes: Gateway Signaling Protocols

H.323

• The H.323 protocol was designed by the ITU-T and initially approved in February 1996. It was developed as a protocol that provides IP networks with traditional telephony functionality.
• In Cisco IP Communications environments, H.323 is widely used with gateways, gatekeepers, and third-party H.323 clients, especially video terminals. You configure connections between devices using static destination IP addresses.
• Because H.323 is a peer-to-peer protocol, H.323 gateways are never controlled by Cisco Unified CallManager. Therefore, H.323 gateways are never registered at the Cisco Unified CallManager. Only the IP address is seen by the Cisco Unified CallManager to confirm that communications is possible.

MGCP

• MGCP is a plain-text protocol used by call-control devices to manage IP telephony gateways. MGCP was defined under RFC 2705, which was updated by RFC 3660, and superseded by RFC 3435, which was updated by RFC 3661.
• is a client/server protocol that allows a call agent (such as Cisco Unified CallManager) to take control of a specific voice port on a gateway.
• With this protocol, the Cisco Unified CallManager knows of and controls individual voice ports on the gateway. It allows complete control of the dial plan from Cisco Unified CallManager, and gives CallManager per-port control of connections to the PSTN, legacy PBX, voice mail systems, POTS phones, and so forth.
• This is implemented with use of a series of plain-text commands sent over User Datagram Protocol (UDP) port 2427 between the Cisco Unified CallManager and the gateway.
• It is important to note that for an MGCP interaction to take place with Cisco Unified CallManager, the gateway must have Cisco Unified CallManager support.
• A PRI and BRI backhaul is an internal interface between the call agent (such as Cisco Unified CallManager) and Cisco gateways. It is a separate channel for backhauling signaling information. A PRI backhaul forwards PRI Layer 3 (Q.931) signaling information via a TCP connection.

SIP

• SIP is a protocol developed by the Internet Engineering Task Force (IETF) Multiparty Multimedia Session Control (MMUSIC) Working Group as an alternative to H.323.
• SIP features are compliant with IETF RFC 2543, published in March 1999, RFC 3261, published in June 2002, and RFC 3665, published in December 2003.
• Because it is a common standard based on the logic of the World Wide Web and very simple to implement; SIP is widely used with gateways and proxy servers within service provider networks for internal and end-customer signaling.
• SIP is a peer-to-peer protocol where user agents (UAs) initiate sessions, like H.323. But unlike H.323, SIP uses ASCII-text-based messages to communicate. Therefore, you can implement and troubleshoot it very easily, and analyze the incoming signaling traffic content very simply.
• Because SIP is a peer-to-peer protocol, the Cisco Unified CallManager does not control SIP devices, and SIP devices do not register with Cisco Unified CallManager. As with H.323 gateways, only the IP address is available on Cisco Unified CallManager to confirm that the communications between the Cisco Unified CallManager and the SIP voice gateway is possible.

SCCP (Skinny Call Control Protocol)

• SCCP is a Cisco proprietary protocol that is used for the communications between Cisco Unified CallManager and terminal endpoints (based on Selsius prorocol, that was bought by Cisco).
• SCCP is a stimulus protocol, meaning any event (such as on-hook, off-hook, buttons pressed, and so on) causes a message to be sent to the Cisco Unified CallManager.
• The Cisco Unified CallManager then sends specific instructions back to the device to tell it what to do about the event.
• Therefore, each press on a phone button causes data traffic between the Cisco Unified CallManager and the terminal endpoint. SCCP is widely used with Cisco IP phones.
• The major advantage of SCCP within Cisco Unified CallManager networks is its proprietary nature, which allows you to make quick changes to the protocol and add features and functionality.

Signaling Protocols Comparison (H.323, MGCP, SIP, SCCP):

H.323

• The H.323 protocol suite is a peer-to-peer protocol. The necessary gateway configuration is relatively complex, because you need to define the dial plan and route patterns directly on the gateway.
• Examples of H.323-capable devices are the Cisco VG224 Analog Phone Gateway and the Cisco 2600XM Series, Cisco 2800 Series, 3700 Series, and 3800 Series routers.
• The H.323 protocol is responsible for the entire signaling between the Cisco Unified CallManager cluster and the gateway. The ISDN protocols, Q.921 and Q.931, are only used on the ISDN link to the PSTN.
  • NOTES:
    • Q.921 – Also referred to as LAPD (Link Access Protocol – D Channel) and a close cousin of HDLC, is the Data Link protocol used over ISDN’s D channel. [ Reference: http://www.freesoft.org/CIE/Topics/125.htm ]
    • Q.931 -  is ISDN’s connection control protocol, roughly comparable to TCP in the Internet protocol stack. Q.931 doesn’t provide flow control or perform retransmission, since the underlying layers are assumed to be reliable and the circuit-oriented nature of ISDN allocates bandwidth in fixed increments of 64 kbps. Q.931 does manage connection setup and breakdown. Like TCP, Q.931 documents both the protocol itself and a protocol state machine. [ Reference: http://www.freesoft.org/CIE/Topics/126.htm ]

MGCP

• The MGCP protocol is based on a client/server architecture.
• That simplifies the configuration because the dial plan and route patterns are defined directly on the Cisco Unified CallManager within the cluster.
• Examples of MGCP-capable devices are the Cisco VG224 Analog Phone Gateway and the Cisco 2600XM Series, 2800 Series, 3700 Series, and 3800 Series routers. Non-IOS MGCP gateways include the Cisco Catalyst 6608-E1 and Catalyst 6608-T1.
• MGCP is used to manage the gateway. All ISDN Layer 3 information is backhauled to the Cisco Unified CallManager. Only the ISDN Layer 2 information (Q.921) is terminated on the gateway.

  

SIP

• SIP is a peer-to-peer protocol.
• The configuration that is necessary for the gateway is relatively complex because the dial plan and route patterns need to be defined directly on the gateway.
• Examples of SIP-capable devices are the Cisco 2800 Series and 3800 Series routers.
• The SIP protocol is responsible for the entire signaling between the Cisco Unified CallManager cluster and the gateway. The ISDN protocols, Q.921 and Q.931, are only used on the ISDN link to the PSTN.

SCCP

• SCCP works in a client/server architecture in the same way as MGCP does. Therefore, it simplifies the configuration of SCCP devices such as Cisco IP phones and Cisco ATA 180 Series and VG200 Series FXS gateways.
• SCCP is used on Cisco VG224 and VG248 analog phone gateways.
• ATA’s enable communications between Cisco Unified CallManager and the gateway.
• The gateway then uses standard analog signaling to the analog device connected to the FXS port. Recent versions of Cisco IOS voice gateways, for example, the 2800 series, also support SCCP controlled FXS ports.

IP-to-IP Gateways

• Cisco Multiservice IP-to-IP Gateways (IPIPGWs) are the next-generation gateways within unified IP communications networks.
• They facilitate connectivity between independent VoIP networks by enabling VoIP and videoconferencing calls from one IP network to another.
• The Cisco Multiservice IPIPGW performs most of the functions of a PSTN-to-IP gateway, but typically joins two IP call legs, rather than a PSTN and an IP call leg.
• Media packets (RTP packets) can flow either through the gateway (thus hiding the networks from each other) or around the gateway, if configured to do so.
• Because the Cisco Multiservice IPIPGW is usually used to interconnect two independent networks like service provider networks or an enterprise network to a VoIP service provider, the gateways are generally configured to terminate the RTP. That makes it possible to have a single point of contact between those two networks, which leads to more security between those networks because the IP-to-IP gateway then functions as a proxy for signaling and voice traffic.
• The Cisco Multiservice IPIPGW also allows the use of two different protocols on both sides. That makes interconnections between two different networks easier and allows simple upgrades from PSTN links to IP links because it doesn’t matter which protocols are already used within the existing network.

Download: [ PDF copy here ]

References:

• Cisco Systems, Inc. Products & Services. http://www.cisco.com/en/US/products/sw/voicesw/index.html.
• http://www.freesoft.org/CIE/Topics/125.htm
• http://www.freesoft.org/CIE/Topics/126.htm
• Cisco PEC (requires a valid account)

Filed under: » CCVP, » Gateway, » Review Notes, » Voice

Reference: H.323 and SIP Comparison

Found this great comparison between this two popular VoIP protocols. This is a great resource to fully understand each other, by doing a comparison. Credit given to “Packetizer“.

Reference: http://www.packetizer.com/ipmc/h323_vs_sip/

Filed under: » CCVP, » Gateway, » Review Notes, » Tutorial, » Voice

Review Notes: Gateway Deployments

A voice gateway allows terminals of one type, such as H.323, to communicate with terminals of another type, such as a PBX, by converting protocols.Gateways connect a company network to the PSTN, a PBX, or individual analog devices such as a phone or fax.

Types of Cisco access gateways:

• Analog Gateways:
  • Analog Station – gateways that connect an IP telephony network to plain old telephone service (POTS). They provide Foreign Exchange Station (FXS) ports to connect analog telephones, interactive voice response (IVR) systems, fax machines, PBX systems, and voice-mail systems.
  • Analog Trunk – gateways that connect an IP telephony network to the PSTN central office (CO) or a PBX. They provide Foreign Exchange Office (FXO) ports for PSTN or PBX access and recEive and transMit (E&M) ports for analog trunk connection to a legacy PBX. To minimize any answer and disconnect supervision issues, use digital gateways whenever possible. Analog direct inward dialing (DID) is also available for PSTN connectivity.

• Digital Gateways: Cisco access digital trunk gateways connect an IP telephony network to the PSTN or to a PBX via digital trunks, such as PRI common channel signaling (CCS), BRI, and T1 or E1 channel associated signaling (CAS). Digital T1 PRI trunks may also connect to certain legacy voice-mail systems.

IP telephony gateways should meet these core feature requirements:

• Gateway protocol support: Gateways support H.323, Media Gateway Control Protocol (MGCP), session initiation protocol (SIP), and Skinny Client Control Protocol (SCCP). H.323 and SIP gateways do not need a call control agent. MGCP and SCCP are streamlined protocols that only work on a network in which a call agent such as a Cisco Unified CallManager is present.

• Advanced gateway functionality:
  • Dual tone multifrequency (DTMF) relay capabilities: Each digit dialed with tone dialing is assigned a unique pair of frequencies. Voice compression of these tones with a low bit-rate codec can cause DTMF signal loss or distortion. Therefore, DTMF tones are separated from the voice bearer stream and sent as signaling indications through the gateway protocol (H.323, SCCP, or MGCP) signaling channel instead.
  • Supplementary services support: These services provide user functions such as hold, transfer, and conferencing, and are considered to be fundamental requirements of any voice installation.
  • Work with redundant Cisco Unified CallManagers: The gateways must support the ability to rehome to a secondary Cisco Unified CallManager in the event of a primary Cisco Unified CallManager failure.
  • Call survivability in Cisco Unified CallManager: The voice gateway preserves the Real-Time Transport Protocol (RTP) bearer stream (the voice conversation) between two IP endpoints when the Cisco Unified CallManager to which the endpoint is registered is no longer accessible.
  • Q Signaling (QSIG) support: QSIG is becoming the standard for PBX interoperability in Europe and North America. With QSIG, the Cisco voice packet network appears to PBXs as a distributed transit PBX that can establish calls to any PBX or other telephony endpoint served by a Cisco gateway, including non-QSIG endpoints. (For interoperability purposes)
  • Fax and modem support: Fax over IP enables interoperability of traditional analog fax machines with IP telephony networks. The fax image is converted from an analog signal and is transmitted as digital data over the packet network.

• Cisco Unified CallManager Release 3.1 and later supports H.323 and MGCP gateway protocols.
• Cisco Unified CallManager Release 4.0 and later also supports SIP.
• Cisco IP phones use SCCP, which is a lighter-weight protocol. SCCP uses a client/server model, while H.323 is a peer-to-peer model. MGCP also follows a client/server model.
• Protocol selection depends on site-specific requirements and the installed base of equipment. For example, most remote branch locations have Cisco 2600XM Series or 3700 Series multiservice routers installed. These routers support H.323 and MGCP 0.1 with Cisco IOS Release 12.2(11)T and Cisco Unified CallManager Release 3.1 or later. For gateway configuration, you might prefer MGCP to H.323 due to simpler configuration or, with older IOS versions, due to support for call survivability during a Cisco Unified CallManager failover from a primary to a secondary Cisco Unified CallManager. Failover with H.323 is only supported with Cisco Unified CallManager 4.1 or later and Cisco IOS Release 12.4(4)XC or later. On the other hand, you might prefer H.323 over MGCP because of the robustness of the interfaces supported.
• The Simplified Message Desk Interface (SMDI) is a standard for integrating voice mail systems to PBXs or Centrex systems. Connecting to a voice-mail system via SMDI and using either analog FXS or digital T1 PRI requires either SCCP or MGCP protocol because H.323 devices do not identify the specific line being used from a group of ports. The use of H.323 gateways for this purpose means the Cisco Messaging Interface cannot correctly correlate the SMDI information with the actual port or channel being used for an incoming call.

“He who wrestles with us strengthens our nerves and sharpens our skill. Our antagonist is our helper” – Edmund Burke, 1729-1797

Download: [ PDF copy here ]

Reference: Cisco PEC (requires a valid account)

Filed under: » CCVP, » Gatekeeper, » Gateway, » Review Notes, » Voice