OVERVIEWOVERVIEW

CALL SCENARIOCALL SCENARIO

THE CISCO Call Manager THE CISCO Call Manager ExpressExpressOPERATION:

• IT tracks all active VOIP | POTS components like phones , gateways , bridges and trans-coding resources

• Configure CLI commands for Cisco IP communicator using the SCCP

• Configure CLI Dial-peer commands for Sjphone using the SIP

• Configure CLI Dial-peer commands for e911 services using POTS

• Provides end to end connectivity between any soft phone using either SCCP or SIP

THE CISCO Call Manager THE CISCO Call Manager ExpressExpressTHE BASIC’s of VOIP DIAL-PEERS:

Router(config)# dial-peer voice number voip

Router(config-dial-peer)# destination-pattern string

Router(config-dial-peer)# session target {ipv4:destination-address | dns:[] host-name}

Router(config-dial-peer)# dtmf-relay [cisco-rtp] [h245-signal] [h245-alphanumeric]

Router(config-dial-peer)# session protocol { sipv1 |sipv2 |cisco}

• Dual tone multi frequency relay is the mechanism where a local VOIP gateway listens for DTMF digits during a call and then sends them uncompressed as either RTP or H.245 packets to the remote VOIP gateway which regenerates DTMF digits and prevents digit loss due to compression

• This in-band uses a special payload type identifier [PTI] in the RTP header of the voice media stream to distinguish digits from the DTMF pad from actual voice communication

THE CISCO Call Manager THE CISCO Call Manager ExpressExpressTHE BASIC’s of IOS TELEPHONEY :

• DHCP service

• IP source address

• SKINNY communication port

• Number of IP phones [max 8]

• Dual-line extensions

• Phone language

• Call Progress Tone

• First extension number

• Direct Inward Dial

• Call forward voice mail service

THE BASIC’s of POTS DIAL-PEERS:

Router(config)# dial-peer

voice number pots

Router(config-dial-peer)#

destination-pattern string

Router(config-dial-peer)# port

location

THE CISCO IP COMMUNICATORTHE CISCO IP COMMUNICATOR

•The IP phone acts as a dumb terminal while the CME is responsible for the entire setup and the tear down of the call.

• The IP phone registers its name, device type and IP address with the CME and then provides its IP port number on which it will receive and send media messages .

• The CME assigns the soft-keys corresponding to each event that takes place during the call scenario like off hook, new call, redial, hold, etc.

• On occurrence of any of the events the CME sends soft-key event messages to the phone and sends periodic call status info messages.

THE SJ SIP SOFT - PHONETHE SJ SIP SOFT - PHONE

• SJ phone is a SIP soft-phone which does not require to be registered with a server if it is being accessed within the LAN [ it is its own user-agent client and user-agent server]

• Integrates with the Microsoft Loopback Driver to receive, parse and translate SIP messages and represent the SIP call scenario

• Supports 1. SYMBIAN OS

2. MAC OS

3. WINDOWS OS

SKINNY – SIP CALL SETUP AND SKINNY – SIP CALL SETUP AND TEARDOWNTEARDOWN

SKINNY – SIP CALL SETUP AND SKINNY – SIP CALL SETUP AND TEARDOWNTEARDOWN

SKINNY – SIP CALL SETUP AND SKINNY – SIP CALL SETUP AND TEARDOWNTEARDOWN

SKINNY – SIP CALL SETUP AND SKINNY – SIP CALL SETUP AND TEARDOWNTEARDOWN

Part- II:Part- II: Development of basic Development of basic SoftPhoneSoftPhoneFunctionality:

•Basic SIP softphone client (like k-lite,sjphone etc)

•Can Register with any SIP server

•Can receive/make calls from/to SIP enabled phones.

•Audio codec used: G.729

•Can add/modify program to achieve more functionality like call forwarding, conferencing, video chat and many more.

Programming Facts for this project:

•Operating System: Win Vista

•portSIP SDK for Microsoft .NET

•Programming Language: C#

•IDE: Microsoft Visual Studio 2008

•Couldn’t have been really possible without help of portSIP SDK documentation.

PortSIP SDK for Microsoft .NETPortSIP SDK for Microsoft .NET

Basic Features:•Support platforms: Windows 2000/XP/2003/Vista, Windows Mobile 5/6, Nokia S60 3rd FP2.

•Support servers: Cisco CallManager, Open SER,  SER, Asterisk, Portaone, Radvision, Nortel, Rainbow, Avaya and other SIP Platforms.

•Audio call: G.711 aLaw/uLaw, GSM, iLBC, G723.1, G729.

•Video call: H263, H263-1998, H264.

•Call transfer: Attended transfer, Blind transfer. Call forwading, Call hold, mute speaker, mute microphone

•IM Support: SIMPLE(Presence, Subscribe, Pager message) and XMPP.

•For full features please see http://www.portsip.com/features.htm

Architecture:•SIP SDK for Microsoft.Net. It is easy to use with c# language once it is understood.

•Contains 3 main libraries

• DeviceManagerLibV4

• PortSIPCoreLibV4

• PortXMPPLibV4

•DeviceManagerLibV:

• Its easy to select audio/video devices on a computer for VOIP communication, using various functions which are written in this library

•PortSIPCoreLibV4

• This library implements the core Session Initiation Protocol stack.

• It has rich set of functions and events

for SIP.

•PortXMPPLibV4• It has implementation of XMPP protocol

for Instant Messaging.

Soft-phone ClientSoft-phone Client

Welcome Screen:•You need to enter your username, password, proxy server name, domain name of proxy server, proxy server port number (If any).

•You must have account for any SIP server to use this softphone.

•We used www.voxalot.com username/password for our testing purpose.

•If username/password is correct, it will try to register it on SIP proxy server you provided.

•If registration is successful, you can make/receive calls from SIP enabled devices.

•To authentication and registration we used initialize() and registerServer() functions from portSIP SDK.

Softphone Client: cntdSoftphone Client: cntdPhone Screen:•Once you are successful authenticated and registered, you will see this screen.

•You can make call or received calls from sip enabled devices now.

•Here we are trying to call ‘408334’ which is also a registered number of voxalot.

•You can see the log of the various events occuring.

•Same way it will give you notification for incoming call.

•We have used following main functions and events from portSIP SDK to make this achieved.

•Methods• Call(ref string callTo,hasSDP) : To place a call.

Returns sessionid

• answerCall(int sessionID) : To answer an incoming call.

• rejectCall() and terminateCall() : To reject/end a call

•Events:• inviteTrying() : when call is trying

• inviteRinging() when phone is ringing

• inviteAnswered() when phone is answered by a person

• inviteIncoming() when there is an incoming call

• inviteClosed(() when person disconnects the call

REFRENCES & AREA OF REFRENCES & AREA OF CONTRIBUTIONCONTRIBUTION• http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/srnd/design/guide/dialplan.html

• http://www-europe.cisco.com/univercd/cc/td/doc/product/software/ios122/122sup/122csum/csum3/122cvvf/vsf_r.htm#1736728

• http://www.cisco.com/en/US/docs/voice_ip_comm/cusrst/admin/srst/configuration/guide/srstsa.html

•www.portsip.com

CHINMAY PADHYECME – IP COMM. – SJ PHONE

SIMULATION

NEHA SHARMASKINNY – SIP CALL FLOW

ANALYSIS

VAIBHAV KULKARNIDEVELOPMENT OF SIP SOFT-

PHONE