VOIP Archives - CEM KAYA

Multi-site PBX Connectivity in ACME SBC

Multi-site PBX Connectivity in ACME SBC : Consider the scenario where the Enterprise has multiple branch offices, each with one or more PBXs deployed. Should there be an ACME Net-Net SD on the network edge of each of the branch offices, then the Basic PBX Connectivity approach in the section above should be used at each office location.
However, should the ACME Net-Net SD be deployed in a data center as an aggregation point for many or all branch offices, then a multi-realm approach is recommended.
Typically, routing per branch office would be on a separate VLAN id. There is a one-to-one correlation of VLAN id and realm. Additionally, there is a one-to-one correlation of realm and sip-interface. Hence, the simplest model of multi-site PBX connectivity would require the Net-Net SD be configured with as many sip-interface IP addresses as […]

ACME, SBC, VOIP|

Topology Hiding in ACME SBC

SIP Manipulation

You can do almost any SIP header manipulation in SIP Manipulation section in ACME SBC. I will show how to do topology hiding in Acme SBC with Header Manipulation Rules (HMRs).

HMR_manip1

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ACME, SBC, VOIP|

ISDN User Part – ISUP

The ISDN User Part – ISUP is responsible for setting up and releasing trunks used for inter-exchange calls. As its name implies, ISUP was created to provide core network signaling that is compatible with ISDN access signaling. The combination of ISDN access signaling and ISUP network signaling provides an end-to-end transport mechanism for signaling data between subscribers. Today, the use of ISUP in the network has far exceeded the use of ISDN on the access side. ISUP provides signaling for both non-ISDN and ISDN traffic; in fact, the majority of ISUP-signaled traffic currently originates from analog access signaling, like that used by basic telephone service phones.

The primary benefits of ISUP are its speed, increased signaling bandwidth, and standardization of message exchange. Providing faster call setup times than Channel Associated Signaling (CAS), it ultimately uses trunk resources more effectively. The […]

SS7, VOIP|

Message Transfer Part 3 – MTP3

Level 3 of the Message Transfer Part (Message Transfer Part 3 – MTP3) resides at layer 3 of the OSI model and performs the SS7 protocol’s network functions. The primary purpose of this protocol level is to route messages between SS7 network nodes in a reliable manner. This responsibility is divided into two categories:

Signaling Message Handling (SMH)
Signaling Network Management (SNM)

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SS7, VOIP|

Message Transfer Part 2 – MTP2

Message Transfer Part 2 – MTP2 corresponds to OSI Layer 2 (the data link layer) and as such is the lowest protocol in the stack. Sitting on the physical layer, it provides a reliable means of transfer for signaling information between two directly connected signaling points (SPs), ensuring that the signaling information is delivered in sequence and error-free.

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SS7, VOIP|

ACME SBC Functions & Features

Security :

SBC DoS/DDoS protection
- Protect SBC from DoS/DDoS attack and other malicious attacks
- Protect SBC from non-malicious overloads
- Allow trusted/authenticated users access while under DoS attack
- Dynamically accept or reject traffic based on device behavior
Access control
- Filter specific devices or whole networks on a per application basis
- Permit access to known devices or networks
- Permit access to from authorized/registered users; permit or deny access to mask users
- Dynamically accept or reject traffic based on device behavior
- Accept media only for authorized sessions
Topology hiding & privacy
- Hide core topology to prevent directed attacks and preserve confidentiality
- Mask user information for privacy and confidentality
- Protect users and service provider infrastracture from eavesdroppers, identity thieves and fraud
- Secure L2 and L3 VPN customers by maintaining security isolation between VPNs ; support inter-VPN sessions
- Support for inter-VPN sessions; monitor media for […]

ACME, SBC, VOIP|

Video over IP QoS (Network Monitoring of MPEG over a IP Transport)

This presentation on monitoring MPEG Transport Streams over IP networks describes test methodology used to ensure superior Quality of Service (QoS) in video. The webinar covers how to use cross layer correlation capabilities when monitoring, to easily detect key problems when they occur when delivering VoIP services. Also covered are the differences between IP video amd voice and data, and how to ensure a high-quality video viewing experience.

Speaker: Steve Holmes

Time: 60 minutes

IP TV, VOIP|

Small Form-Factor Pluggable Transceiver – SFP

A small Form-Factor Pluggable Transceiver – SFP transceiver is a compact, hot-swappable, input/output transceiver used in data communication and telecommunications networks. SFP interfaces between communication devices like switches, routers and fiber optic cables, and performs conversions between optical and electrical signals. SFP transceivers support communications standards including synchronous optical networking (SONET)/synchronous digital hierarchy (SDH), gigabit ethernet and fiber channel. They also allow the transport of fast Ethernet and gigabit Ethernet LAN packets over time-division-multiplexing-based WANs, as well as the transmission of E1/T1 streams over packet-switched networks.

Cisco Fast Ethernet SFP Modules

SFP is also called a mini gigabit interface converter (GBIC) because its function is similar to the GBIC transceiver but with much smaller dimensions.

Compact SFP Deployment with Cisco Catalyst 4500

VOIP|

Configuring a Cisco Router
Gallery
Configuring a Cisco Router

CISCO, NETWORK, VOIP

Configuring a Cisco Router

In the following sections, I’ll introduce the commands used to do basic router configuration.
You’ll use these commands (or should use them) on every router while you re configuring a Cisco router.

Setting a Router Hostname

Router#config t
Router(config)#hostname Kaya
Kaya(config)#

Setting Passwords

Here’s an example of setting all your passwords and then encrypting them in the plain configuration file:

Kaya#config t
Kaya(config)#line aux ?
<0-0> First Line number
Kaya(config)#line aux 0
Kaya(config-line)#password aux
Kaya(config-line)#login
Kaya(config-line)#line con 0
Kaya(config-line)#password console
Kaya(config-line)#login
Kaya(config-line)#exit
Kaya(config)#line vty 0 ?
<1-1180> Last Line number
<cr>
Kaya(config)#line vty 0 1180

CISCO, NETWORK, VOIP|

H323 Signalling Messages

— I will complete the whole signalling soon —

H225/Q931 CALL SETUP

SETUP : source&dest address,port

CALL PROCEEDING

ALERTING : ringback tone

The called telephone starts ringing and this is signalled to the other party with the Q.931/H.225.0 message Alerting.

H245 NEGOTIATION AND VOICE PATH SETUP

CONNECT (Q931/H225) : create h245 channel

The called party picks up the handset and the endpoint can signal the call has been accepted. This is done by sending the Q.931/H.225.0 message Connect. At this point, the parties will need to negotiate parameters for audio (and optionally video) channels. The protocol H.245 is used for this negotiation. In the default case, H.245 will need another TCP channel, so the endpoint 122 will insert its H.245 address (the IP address and port where it listens for H.245 connections) into […]

H323, VOIP|