Transcript VoIP_Basics - Alpesh Parmar`s Site

VoIP Basics
-Created by VoIP-Support Team
YOU Telecom India Pvt. Ltd.
1) What is VoIP ?
Voice Over Internet Protocol converts voice signals from your telephone into a digital
signal that travels over internet then converts it back to voice signals at the other end so
that you can speak to anyone having a regular phone no.
2) How does VoIP work?
Customer will dial any no. using the phone connected to VoIP device & that call will
be forwarded to our soft-switch using our network. Our soft-switch will forward the call
to termination-partner using internet. Termination-partner will divert the call to PSTN
in turn & the call will get connected
3) Who can provide VoIP service ?
VoIP services can be provided by an ISP who had upgraded their ISP license with ITSP
(Internet Telephony Service Provider) license
4) Major VoIP termination partners
-Net2Phone
-Vonage
-Net4India
5) Protocols used in VoIP
-H.323
-SIP (Session Initiation Protocol)
6) Codecs used in VoIP
-Codes are mainly used for analog to digital & digital to analog conversion. We usually
Support G.723.1,G.729,G.711
7) Different Types of Calling in VoIP
a) PC to PC
Cable
Modem
YOU Network
VoIP Switch
PC
PC
Internet
b) PC to Phone
PC
YOU
Network
Cable
Modem
VoIP Switch
Termination
Partner
PSTN
Internet
Phone
c) Phone to Phone
Phone
Phone
VoIP Device
Cable
Modem
PSTN
8) Entities involved in VoIP calling
-ISP
-VoIP Service Provider
Termination
Partner
YOU
Network
VoIP Switch
Internet
9) Requirements for PC to Phone Calling(using softphone)
-PC with windows based operating system
-Internet on PC
-Full duplex sound card
-Speakers or headphone and Microphone
-YOU Softphone dialer
-YOU VoIP account details (User ID or Password)
10) Requirements for Phone to Phone Calling
-Analog Phone Instrument
-VoIP Device or IP phone
-Internet connection
-YOU VoIP account
-Currently we are supporting SIP for VoIP services so VoIP device should be SIP based
11) VoIP Devices we are using
-Grandstream
-Cisco ATA 186
-Linksys PAP2
-Dlink DGP-202SP
-Audio codes 4,8 & 24 ports device
-Any industry standard device which can support SIP
12) Block Diagram
13) Basic Call Flow
1.
2.
3.
4.
5.
6.
7.
We need to connect the IP phone behind cable modem
IP Phone will establish a PPPoE/Static connection using cable modem with CMTS
After the connectivity device will send register request to proxy
Proxy will send back 200OK in response
After successful registration, you will get dial tone
Once you get the dial tone you can start making calls
Whenever you make any call, IP Phone will send an invite to proxy & proxy will check in its
database that the sufficient balance is available in the account or not. If balance is there then
call will be allowed. Proxy will reply back with 200OK
8. After above step proxy will forward trying to IP phone
9. Proxy will forward the invite for our carrier
10. Carrier will provide ringing
11. Then proxy will forward ringing to IP phone
12. If remote peaks up then carrier will provide 200OK
13. Proxy will forward 200OK to IP phone
14. IP phone will send ACK to proxy & proxy will forward ACK to carrier
15. At this step 2 two way media channel will be established between IP phone & carrier
16. Finally IP phone will disconnect the call & will send BYE message to proxy
17. Proxy will reply back with trying & forward the BYE message to carrier
18. Carrier will reply back with 200OK to proxy which will be forwarded to IP phone
13) Troubleshooting
1) Device based troubleshooting
-No dial tone- Check for connectivity between modem & IP phone
- Check for CPE behind modem
- Modem is online or not
- IP phone configuration
-Engage tone - Check if the no. dialed is correct
- Try changing the telephone instrument
- Destination end problem
-Voice clarity - Check for latency form CMTS to IP phone & CMTS to carrier
- Check for SNR
2) Softphone based troubleshooting
-Invalid UID/Password- Check if the UID/Password is correct
-Call snaped/ Insufficient Fund – User doesn't have sufficient fund
-Number not accessible - Check whether the no. dialed is proper or not
-Call authorization failed - Check the validity of user’s account
-Unable to initialize - Check if the user connected to internet & no firewall is
installed on his pc. Also his ISP is not blocking TCP port
4343 & 4321 in case if he is using third party net connection
-Fatal error – OS related issue, try to reinstall the dialer
14) In case of any problem you can always contact us at
• E-Mail ID: [email protected]
• Land-line No.: +91-261-2789500 Extn.: 1650,1651,1652
• Mobile No.: +91-98790-00854 or +91-98251-00189
• VoIP No.: 9-261-16500