What is VOIP?

WHAT I LEARNT TODAY: TIDBITS

Voice over IP converts voice into digital signals that travel over the internet.

TDM - means traditional time-division multiplexing which is what traditional PSTN networks use. A lot of bandwidth is lost here when there is no voice traffic.

PACKET TELEPHONY - This is packet-based voip transmission and is able to multiplex voice traffic side by side with data traffic - more efficient.

COMPONENTS OF A PACKET VOICE NETWORK:
-ip phones
-gatekeeper
-gateway
-multipoint control unit
-call agent
-applicationservers
-video conferencing station

VOIP IMPLEMENTATION - REQUIRED FUNCTIONALITY:

1. SIGNALLING

- the ability to generate and exchange control information that will be used to establish, maintain and release connections between two endpoints

Voice signalling requires the ability to provide supervisory address and alerting functionality between nodes.

PSTN uses SS7 to transport control messages. SS7 uses OOB signalling(exchange of call control info in a separate dedicated channel)

2. DATABASE SERVICES

- the ability to query databases - this is needed to detremine if a call can be placed or not. (Billing info, CNAM delivery, toll-free dtabase services, calling card sevices, etc)

3. BEARER CHANNELS:

- carry voice calls
-PSTN network: connect/disconnect messages carried by SS7
-IP network: carried by SIP, H.323, H.248, MGCP

4. CODECS:

- provide coding/decoding translation between analog and digital facilities (e.g. G.729)

SIGNALING PROTOCOLS: TYPES
1. P2P:
-end devices of gateways contain the intelligence to initiate/terminate calls interprete call control messages.
-SIP, H.323

2. CLIENT/SERVER ARCHI
- no call control intel but can send/receive event notifications to the server (call agent./)
- H.248, SCCP, MGCP


VOIP SIGNALING PROTOCOLS:

1. H.323
Specifies the components/protocols/procedures that provide multimedia services over packet networks including IP networks - real-time audio/video/data comms.

Also defines end2end call-signaling.

Five protocols specified by H.323 are:

a. H.225 Call Signaling - used to establish connection between two H.323 endpoints.
b. H.225 registration, admission and status (RAS) - lies between endpoints (terminals and gateways) and gatekeepers. RAS is used to perform registration, admission control, bandwidth changes and status and disengage procedures between endpoints and gatekeepers.
c. H.245 Control Signaling - used to exchange e2e control messages governing the op of the H.323 endpoint. The control messages carry info relating to

- capabilities exchange
- opening and closing of logical channels used to carry media streams
- flow-control messages
- general commands and indicators

c. Audio Codec - encodes the audio signal from the mic for transmission on the transmitting H.323 terminal and decodes the received audio code sent from the receiving H.323 terminal to the speaker.

Audio is the minimum standard provided by H.323, hence all H.323 terminals must have at least one audio codec support as specified by the ITU-T G.711 recomm (audio codec at 64kbps).

Some other audio codec recomms that may also be supported:

- G.722 (64, 56, and 48 kbps)
- G.723.1 (5.3 and 6.3 kbps)
- G.728 (16kbps)
- G.729 (8 kbps)

D. Video Codecs - Like the audio one but having to do with video.
H.323 must conform with the ITU-T H.261 recommendation.

Remember, H.323 is a p2p protocol so H.323 gateways are not registered with CUCM as an endpoint, it needs an IP add to be configured in CUCM for comm to take place.

The necessary gateway configs are complex - need to define dial plans and route patterns directly to the gateway. H.323 does all the signaling between the CUCM and the gateway

Some H.323 capable devices are:

- Cisco VG224 Analog Phone Gateway
- Cisco 2600XM Series Multiservice Routers
- Cisco 2800 Series Integrated Services Routers
- Cisco 3700 Series Multiservice Access Routers
- Cisco 3800 Series Integrated Services Routers


2. MGCP
Defines a method for PSTN gateway control or thin device control.
Controls VOIP gateways connected to external call control devices (call agents).
Provides signaling ability for cheap edge devices that may not have full voice-signaling protocol, such as H.323, implemented.

MGCP is a client/server protocol and is built on centralized control archi.

All dial plan info reside on a separate CALL AGENT, which controls ports on the gateway and performs call control.

The gateway does media translation between the PSTN and VOIP networks for external calls. CUCM functions as CALL AGENT in a Cisco network.

MGCP is plain-text.

MGCP allows complete control of the dial plan from CUCM which it also gives per port control of connections to the PSTN, legacy private branch exchange (PBX), voice-mail systems and POTS, etc. This control is implemented by series of plain-text commands sent between CUCM and the gateway over UDP port 2427. the gateway must have CUCM support.

A PRI and BRI backhaul is an internal interface between the call agent eg CUCM and the Cisco gateways.

A PRI backhaul forwards PRI Layer 3 (Q.931) signaling info via a TCP con.

A call agent must always be available for a MGCP gateway, which is itself relatively easy to configure since the call agent has all the call-routing intelligence.

Cisco MGCP gateways can use SRST and MGCP fallback to allow the H.323 prot to take over and provide local call routing in the absence of a CUCM. Dial peers must then be configured on the gateway for use by H.323.

MGCP is relatively easy to config due to its client/server archi. The dial plan and route patterns are defined directly on the CUCM within the cluster.

MGCP is used to manage the gateway. All ISDN Layer 3 (Q.931)info is backhauled to the CUCM. The ISDN Layer 2 (Q.921) info is terminated on the gateway.

3. SIP

Is P2P
Specifies commands to initiate and tear down calls.
Features security, proxy, tcp, udp.
Works with SAP and SDP to to provide announcements and info about multicast sessions to users on a network.
Defines e2e call-signaling between devices.
ASCII Text-based
Much like HTTP (same transaction request/response model and similar header/response codes.)

Since it is p2p, CUCM does not control SIP devices and SIP devices do not register with CUCM.

Only IP add is used to confirm communication between CUCM and SIP gateway (much like H.323).

The necessary gateway configs are complex - need to define dial plans and route patterns directly to the gateway. SIP does all the signaling between the CUCM and the gateway. The ISDN prots Q.921 and Q.931 are used only on the ISDN link to the PSTN - same as H.323

4. SCCP

Cisco-proprietary.
Used between CUCM and Cisco VOIP phones.
End stations using SCCP are called SKINNY clients (consume less processing overhead).

It is a stimulus prot - any event triggers messages to the CUCM, which then send it detailed instructions on what to do about the events.

Cisco devices using SCCP can coexist with H.323 terminals.

VOIP SERVICE CONSIDERATIONS:

To configure voice on a data network, there has to be low delay,minimal jitter and minimal packet loss. Bandwidth requirements must be met, QoS must be configured to minimize jitter and loss of voice packets.

1. Latency

- Increase Bandwidth
- Choose diff codec type
- Fragment data packets
- Prioritize voice packets

2. Jitter

- use dejitter buffers

3. Bandwidth

- Resolve bandwidth issues by calculating bandwidth requirements including payload , overhead and data.

4. Packet Loss

- design network to minimize congestion
- prioritize voice packets
- use codecs to minimize small amounts of packet loss

5. Reliability

- provide redundancy for hardware, links, and power (UPS)
perform proactive network management

6. Security

- Secure network infrastructure
- secure call-processing systems
- secure endpoints
- secure apps