August 2007 Archives

SIP Trunking Tutorial - SIP-T-SIP for Telephones: IP-to-PSTN via SS7 - RFC 3372

August 25, 2007 12:09 PM | 0 Comments

 This is an introduction to SIP Trunking and connections for telephones and discussion of SIP telephony network with respect to the PSTN-Public Switched Telephone Network via SS7-Signaling System 7. This tutorial will present the various SS7 protocols used in PSTN signaling from a SIP phone. Later, other call processing scenarios will be presented. The following are the steps in the SIP signaling process to complete a call via the PSTN-Public Switched Telephone Network. There are others but this is a start. First, the SIP device sends an INVITE.

 
The GMC-Gateway Media Controller converts IP-Internet Protocol packets to CAS-Channel Associated Signaling TDM-Time Division Multiplexed channels signaling and media transmission with the PSTN.
 
A Q.931-SS7-ISUP-ISDN-Integrated Services Digital Network User Part IAM-Initial Address Message is sent with the Routing label, CIC-Circuit Identification Code and Message Type Code. Examples of Message Type Codes include: Called Number, Calling Number, DPC-Destination Point Code, OPC-Origination Point Code.   As the IAM passes through switching centers circuits (shown in green) are reserved. 
 
Once the IAM-Initial Address Message is processed at the DPC-Destination Point Code (switch), an SS7-ISUP ACM-Address Complete Message is sent to the OPC-Origination Point Code (switch).   The ACM is used also used to indicate the start of billing.
 
Upon off-hook connect, the reserved trunks are activated and an ANM-ANswer Message completes the call setup/connect and communication (conversation) begins.
 
RTP-Realtime Transport Protocol or SRTC-Secure RTP (encrypted media) is the Session Layer protocol used for the voice transmission with UDP-User Datagram Protocol and then IP for transmission.
 
During the call, various Call Progress messages such as CPG-Call ProGress Message, COT-COTinuity Message, SUS-SUSpend Message, RES-RESume Message, FOT-FOrward message Transfer, INR-INformation Request message, INF-INFormation Message and others may be sent.
 
Upon RELease or disconnect (on-hook), a series of SS7-ISUP messages are sent to teardown (release) the trunks, terminate billing, and provide network management and testing for the next call.
 
This presentation is part of the SIP-VoIP Essentials online course available from TCMnet at http://www.tmcnet.com/tmcnet/mkt/nacse/sip-essentials.htm or the version for Channel Partners called NVBE http://www.tmcnet.com/tmcnet/mkt/nacse/.
For access to more than 2,930 other tutorials on technology, go to http://www.techtionary.com

 

 

Dumb POTS to Smart SIP – Introduction to SIP Trunking

August 16, 2007 4:37 PM | 0 Comments

In an IP network, the intelligence is moved to the "edge" in the form of digital phones that support protocols such as SIP-Session Initiation Protocol and others for signaling. RTP-Realtime Transport Protocol and UDP-User Datagram Protocol-IP-Interent Protocol for media transmission. SIP phones generally do not support analog such as FAX.
 
Finding telephone numbers are functions of the Home proxy server designed to provide Location Service for Local users. AOR-Address Of Record is usually the users' public address such as sip:name@company.com or "tel" URI such as tel:303-594-0909.   DNS-Domain Name Service is used to find/route public users and send back a 404 Not Found, if address is not found or 486 if Busy.
 
If the telephone number is NOT found in the DNS-Domain Name Server, an outgoing call will be connected via a Gateway to the PSTN-Public Switched Telephone Network or LD-Long Distance using TDM-Time Division Multiplexed circuit-switched systems (e.g. Class 5 CO-Central Office). 
 
For incoming calls, for example, if both you and another company have SIP-enabled multimedia infrastructures, the conference or individual communications could be conducted directly via the Internet, with no need to reserve conference time or pay for LD service.   If the incoming call is not on the IP network, the incoming call from the PSTN would be converted to IP and routed to the SIP device.
 
This presentation is part of the SIP-VoIP Essentials online course available from TCMnet at http://www.tmcnet.com/tmcnet/mkt/nacse/sip-essentials.htm or the version for Channel Partners called NVBE http://www.tmcnet.com/tmcnet/mkt/nacse/.

SIP Essentials - SIP Hardware Devices

August 13, 2007 12:38 PM | 0 Comments

The nomenclature for devices in a SIP network generally consist of:
UA-User Agent is a logical (not necessarily physical) function that initiates or responds to SIP communications. A UA can be either a client or server that maintains a "session" status in the SIP communications.
UAC-User Agent Client is also a logical and physical function that initiates or responds to SIP communications.
UAS-User Agent Server is a logical and physical function that initiates or responds to SIP communications.
Proxy Server acts like (hence the term proxy) as an intermediate server that is responsible for forwarding SIP requests to the destination UAS-User Agent Server on behalf of the UAC-User Agent Client.
Redirect Server is an optional SIP component that does not route SIP messages. Redirect Server returns a redirect (via 3xx class address change in routing) to UA-User Agent or UAC-User Agent Client for direct routing (SIP is designed for end-to-end signaling without intervention by a server).
Registrar Server is a user agent server that accepts SIP REGISTER requests and updates.
B2BUA-Back-to-Back User Agent acts like a central PBX connection server processing incoming and outgoing messaging between other user agent servers. The B2BUA maintains "answer supervision" (overseeing) the dialog state and participates in all dialog communications.
This presentation is part of the SIP-VoIP Essentials online course available from TCMnet at http://www.tmcnet.com/tmcnet/mkt/nacse/sip-essentials.htm or the version for Channel Partners called NVBE http://www.tmcnet.com/tmcnet/mkt/nacse/.

 

Introduction to VoIP-SIP Security - SSH-Secure SHell

August 4, 2007 10:35 AM | 0 Comments

 

This is intended to be an introduction to SSH-Secure SHell, not an indepth analysis of the topic. Shown here is an example of SSH is in a VoIP-Voice over Internet Protocol. SSH can be used in all kinds of IP situations. One of the key points of SSH is that the connection is established and encrypted (passwords are not passed between devices in plain text) before a user is possible. SSH keys are asymmetric or symmetric. Symmetric keys are used for the session and same (symmetric) key is used to encrypt and decrypt. That is, symmetric keys are derived (created) from one another. Asymmetric uses pairs of Public and Private keys.  
 
Rules are:
- if you encrypt with the private key, it can only be DEcrypted with the public key.
- if you encrypt with the public key, it can only be DEcrypted with the private key.
SSH uses common hash values.
For more information www.openssh.org
Here are a few words on "keys."
- Public key - uses a key (means of "scrambling" or encryption) from a public source but still has its own "grooves" or codes to scramble the content or message. There are many types of encryption such as PGP-Pretty Good Privacy which has 2 to the 1024 (two followed by 24 zeroes) power "grooves" or key combinations, 56-bit, single-key DES encryption, or others that support the much stronger 168-bit, three-key Triple DES-Data Encryption Standard - AES-Advanced Encryption Standard (tunnel) to encrypt files.
- Private key - create your own "skeleton" key which no else knows about.
Here are the processes in SSH:
SSH - 1- Client requests session on Port 22
SSH - 2 - Server sends Public Host Key or SSH1 - Server sends Public Host Key
and Public Server Key
SSH - 3 - When the client receives the Public Keys, it checks (verifies) to see if the Public Key exists in the client-wide (server) or local ssh_known_hosts.
SSH - 4 - If the host exists in either of these files, the host is validated and the host authentication is complete.
 
This presentation is part of the SIP-VoIP Essentials online course available from TCMnet at http://www.tmcnet.com/tmcnet/mkt/nacse/sip-essentials.htm or the version for Channel Partners called NVBE http://www.tmcnet.com/tmcnet/mkt/nacse/.

Join Me at the Channel Partners for Presentation on Convergence Ahead

August 3, 2007 3:06 PM | 0 Comments
Join me at the Channel Partners Conference & Expo for Convergence Ahead
 
A Free Seminar Sponsored by XO Communications
Thursday, September 27, 2007
9:00 – 11:00 AM
Meadowlands Exposition Center
Room E, 2nd Floor
 
Convergence is here! “U.S. SMBs are spending $30B on converged communications and managed services in 2007,” according to AMI Partners.  So, while you’re at the 2007 fall Channel Partners Conference & Expo, spend a couple hours to learn about how Channel Partners can benefit by offering converged, scalable, cost-saving IP-networking solutions that fit any customer’s unique business needs.
 
In this free seminar, you will learn the following from industry experts:
  • The target markets for converged IP solutions.
  • The Total Costs of Ownership (TCO) and benefits of evolving to Voice over IP (VoIP).
  • The 10 steps to successful VoIP implementation.
  • How XO’s IP portfolio, including our newest SIP product, SIP Trunks, will help you grow your IP sales.
For more on the conference go to http://www.channelpartners.phoneplusmag.com/
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