Why SIP is most hot technology in telecom? VoIP and Unified Comunication

VoIP: Voice over IP.

VoIP over business or enterprise class environment important terms used are VOIP server ,VoIP client,CODEC,gateways,lateny QOS,hard phone,soft phone.These are explained with Example below:

VoIP Server: or IP PABX, all traditional phones goes through PABX, data devices transmit real time audio communications use VoIP servers and VoIP gateways are used to connect outside telephone network to VoIP phones using gateways.

VoIP server connected to switch, which connected to router,switch is also connected to client,all communication happens TCP/IP(ethernet network) but if server has Analog port they can also connect to traditional phone directly, communication provide VoIP IP trunk lines (straight over internet) like company called OnSIP,Normal  Phone line can also used if VoIP server has plugs for phone lines.VoIP server replaces PBAX, VoIP server are hosted over internet.(pay for service every month just like gmail)

Avaya ,Luccent ,AT&T ,Nortel : Propertory VoIP service provider

Asterix,SIPx,Switch Fox: Open source VoIP server Providers.(Now avaya and Cisco also joined here through unified communication).

HardPhone Vs Soft Phone:

HardPhone is also computer device which looks like phone(polycom)VoIP telephone which looks like telephone.To configure through web interface. in polycom goto IP address web server which resides over this phone and configure this phone.

SoftPhone are software which lets computer devices make behave like telephone over software.

VoIP server we can create accounts ,by using these user accounts only client can connect to server. you can configure user name and password. and extension on client. User name and password authentication is must.

VoIP account configured over VoIP server validates VoIP client.

VoIP Gateway: helps old PABX telephone system to connect to VoIP server using RJ11 telephone connector to VoIP server Thus old PABX based phones can also send and receive call using VoIP server.VoIP gateway are connected to traditional phone on one end and VoIP server on other.

How to reduce Long distance call cost to remote office using VoIP server?

A VoIP gateway installed in central office.Everyone uses VoIP phone which uses internet connections. Long distance call charges can be reduce. Call gets diverted over T1 line to VoIP server of remote office which connect to VoIP gateway on remote location office thus can connect normal PSTN network locally to connect local telephone lines for local call. Thus gateway help can help call local no. Long distance call comes over internet lines

office like Sears,Jcpenny, Axon ,have facilities distributed across country and world use this method to reduce call rates for long distance call to remote located office across the world.

skype, voip services Gateway specially designed skype voip traffic can be converted normal sip VoIP.

three communication methods possible skype to SIP, SIP to skype,skype to telephone,. etc for all this VoIP gateways are used.

Session Initiation protocol. SIP cisco phone (very costly),polycom, linksys(relatively cheap).SIP is application layer protocol which works over TCP/IP(hence work over TCP,UDP etc.) Why SIP is important? Major manufacturer uses proprietary system/protocol.

if u use Avaya VoIP server/PABX  then u have only use Avaya VoIP phone.Skype uses proprietory protocol if u use skype u proprietory protocol have to use skype protocol built into it.(u have to same server and client). Cisco deals with SIP,Cisco SIP/VoIP phone can connect open free source SIP VoIP server.(Cisco call manager connect to linksys phone,cisco phone to linksys (nortel,avaya use propertery protocol)server.

Codec: Voice traffic is encoded/encapsulate packets . high quality communication then use high bandwidth codec.(determine quality /sound/audio quality of communications).codec determines how much bandwidth is used? VoIP server have built in codec(open source (sound quality good but bandwith utilization is not good)or properitory Codec (sound quality good but bandwidth utilization is good)). $5 per device which suppose google uses 4.5 kbps are high quality.

NETWORK LATENCY or QOS.

QOS:–> telephone + computers over same line then telephone had to given certain minimum bandwith(priorities). Switches and routers you can prioritize traffic for VoIP.

Network Latency: Hosted VoIP server is over internet. Internet network latency should not be very high.

how long it takes for bit to reach from point A to point B.(45 milliseconds average).VoIP (75-100 milliseconds delay).

Unified communication:boundary between telephone and computer is blurring).Softphone on computer why can you have outlook plugin and when u receive mail u can talk to them. So Microsoft acquired Skype. Outlook email can be turned in audio file and vice versa.Outlook click there name and call them software lets u connect to no.(outlook can record them using VoIP server in text u can do google search).

Microsoft+Sype+polycom:

These events highlighted the advantages of using Polycom’s full suite of HD voice, video, telepresence and infrastructure solutions that will natively integrate with the UC solution from Microsoft Lync.

Polycom solutions integrate with Microsoft Lync without gateways, making them easy to deploy, manage, and use. Company offers a complete line of standards-based, fully interoperable voice, video, and telepresence solutions for the Microsoft UC platform, which includes Microsoft Lync, Microsoft Exchange Server, and Microsoft SharePoint Server.

With Microsoft and Polycom, people can now communicate using familiar applications, adding rich voice and video to IM, e-mail, calendars, and more. Polycom voice and video solutions – for Microsoft Lync – support system-wide secure authentication and media encryption, securing calls inside and outside the network. With Microsoft’s Active Directory® solution, people have access to a common directory across Microsoft applications and Polycom solutions, reducing the time and effort required to deploy and manage communications. The joint solution enable life-like communications in enterprise and business-to-business (B2B) environments.

Avaya another major player in voice market announced server to unify communications via SIP in march 2009

Avaya is announcing at VoiceCon Orlando a new server that connects disparate SIP-enabled PABXs into a single system with global dial plans and call routing – a cost-saving first step toward rolling out corporate-wide unified communication.

Avaya Aura Session Manager can enable centralized control of voice, video, messaging, presence and Web applications that the company says can be rolled out quickly to the largest corporations.

Aura architecture can link individual user profiles to sets of communications applications to extend the benefits of unified communications across the business, the company says.

But initially, businesses can use Aura Session Manager to save money by unifying their disparate multi-vendor PABXs. As long as they are SIP capable – and this includes TDM PABX with SIP gateways – they can use SIP trunking to connect all corporate sites. The Aura server can route calls across the network, generating savings on inter-site toll calls.

Aura will work with certain Cisco and Nortel PABXs.

Tied in with Avaya Communication Manager (now Aura Communication Manager) the new Aura Session Manager server can coordinate voice and video features and make them available via SIP across the business network.

Avaya SIP session management to ensure it works with SIP trunking services from AT&T, Orange and Verizon

Avaya Voice Portal software  helps in improve call handling in contact centers after-hours call going to voicemail, Session Manager routes it to a voice portal where the caller can order supplies all using SIP.Scaling up is very easy in VoIP over SIP servers.

Demystifying Telecom network

The telephone system:

PBX not VOIP and unified communication.

how does telephone call get to you? PSTN public switched telephone network. Central offices every single house having telephone line connected to central network (demarc point to telephone pole).

PSTN network of central offices.north American dialing plan NADP.(routing for telephone system)

NADP used telephone no to route you to central offices.

Trunk Lines: No. of trunk call depend on how many trunk u can connect to central office.

40 people 4 trunk/lines  are enough these many call are going at same time.(incomming or outgoing).

each single trunk/line as single no. (Hunt group for all people calling in if 1 busy got to 2 etc..) if call goes it gives 1 no. if not available forwarded to no.2 if 2 busy to 3 etc… till u use all your Hunt group.

1              410 685 2808

2              410 685 2885

3              410 685 2880

4              410 685 2889

for all line calls going out. programe in PBX allow any no for outgoing only.only certain station or lines are allowed.(out calling groups).

PBX:-> extn : 345 to particular no. also to outside no. call routing(permission, Hunt group,call group,out group).

Voice mail is add on to PBX system. Avaya routing AUDIX. (auto attendent AA message reside on voice mail routing on PBX  so u develop rule #1 –> 309 #2 –> 500, etx 800 is AA etc.)

Stations.

Just like your inter comm, call boxes in building,telephone system all are connected to PBX.

Transmission option (analog and digital).(using carrier waves it is transmited,interfreance case to reduce quality of it). Digital(1,0) and digital PBX .so analog and digital cannot talk to each other.fax manchine cannot work with digital PBX.mordern PBX have analog port (for fax etc.)and Digital ports. Just like switch.

Subscribers(users of voice mail system)  not always have to attached with stations.voice mail box have auto attendent with have auto reply for each call. PBX(4 incoming line,9 station/trunk line,2 trunk line for voice mail means only 2 auto attendant).voice mail system  No. of ports means that many no can get auto attended. VOIP is not telephone, its real time data communication over data network with simulated telephone.

Telephone System Call Routing:

T-System can send call to you. hunt group, auto attended, extent clinic, voice mail.

PBX- Everything is related to extension. Voice mail,Intercom,Hunt group, call group,voice boxes also have extension. Auto attend are accessed through  extension. Administrative extension nobody will call.

Call Path: What happen to call goes to certain extension.(programmed 1st ring 3 times,2nd call to voice mail). Call 301 to sales manager rings 3 times–> then goes to his Assistant manger  ring 3 times–>no answer-> call group (all phone on group) ring 3 times–>  goes to emergency no.

Call path: 301–> 1st directly voice mail –>(you can do 50 step call path).

OutCalling: outside people calling all request comes to PBX. Out call go through PBX.Out call in PBX forwarding call to extn (504) to PBX route call to mobile phone.(company do not like out calling).Out calling also consumes a trunk for outgoing call.(single call uses 2 trunk one incoming, one out going).

Trunk Group:

-4 trunk lines i.e. phone no.

1              410 685 2808

2              410 685 2885

3              410 685 2880

4              410 685 2889

Trunk group 1 (1,2) extn 800. business 1 –> auto attendant –> person A  (building A)

Trunk group 2(3,4) Extn. 900. bussiness2 –> auto attendant –> person B  (building A)

Auto Attendant AA:

Extension 800 is auto attendant (configuration inside PABX : business hours , business days (e.g. weekdays).Is  this Extn AA or HG, or ? where does voice comes for AA? PBX routes call to Voice Mail

TG AA1  800, 800 is AA

(VM). AA1-> VM 800 –> (press 1 –>300), (press2 –>400),( press 3 –>500),( press 4 –>600). Time out after each 1 minute.

AA –> 800– press 2 for sales –> (2=900)–> 900–> HG (sales)if call come first go to 301 thne 302 etc..configuration  {(#1,301), (#2,301), (#3,301), (#4,301)}.

Waited Hunt Group:(call HG look for which has picked up more call , it will give call to person which has picked least calls).

Call Groups: (every single phone rings at same time. whoever picks the call get else get dialer tone)

Outgoing call.–>

Out TG1 –> Local call.

Dial 1 then only use –>Out TG2 –> Local + Long distance.

15 digit no. and first 4 (1976) then use TG2 –> to call long distance calls.

443-XXX-XXXX– 10 digit no.–> TG1 or  TG2.

11 digit –>and first digit 1 then  Tg2. press (1) –> long distance.

if 11 digit no use TG2–> Long distance ,(9781) – (1)- (443-xxx-xxxx) long distance call.