VoIP Phones - Use of signaling protocol, call manager, and SIP clients

Q. What is VOIP signaling protocol and what it is for?

A. H.323, MGCP, SIP, etc. all are VoIP signaling protocols, but SIP is most widely accepted VoIP protocol. SIP is responsible for establishing, maintaining, modifying and tearing down the RTP session. However, SIP has no concern with RTP, RTP exchange information is handled by SDP (session description protocol).

Q. What is the meaning of CCVP?

A. CCVP covers all the VoIP signaling protocols, which are used by Cisco in their VoIP servers and clients. CCVP focuses mainly on the configuration part of only own products. It also covers Quality of Service (QoS), which is must for real time applications.

Q. Can you explain about call manager and all other devices used to implement VOIP tech?

A. Call manager is not a very generic term in VoIP. Users select the call manager on the basis of requirement. I believe that you can use the call manager, which includes the proxy and registrar server. Asterisk (http://www.asterisk.org) is the most widely used open source call manager (remember, call manager is also the brand of Cisco VoIP server). With call manager such as asterisk, which you can configure on any PC, you also need SIP clients such as IP Phone, Softphone, ATA, etc.

Q. How can I know IP of an IP phone?

A. Every IP phone has different types of user interface. Please check with the IP-Phone manual that you are using. In general, IP phone displays the currently used IP, subnet mask, etc. somewhere on the LCD display.

Q. Do every Ip phone will have IP by default?

A. Yes, every IP phone has default IP.

Q. If so, is that will be a public IP or private IP?

A. Default IP will always be private one.

Q. What exactly a VOIP server does?

A. VoIP (SIP) server is mainly responsible for the following:

REGISTRAR - With the help of register service of SIP, you (SIP Client) register your current location (IP Address of URL) to the SIP server. You can have multiple bindings with the VoIP server. This helps SIP server locate you at the time of incoming call for you.

PROXY - Proxy server is responsible for sending SIP messages to the caller on behalf of you. Proxy interacts with REGISTRAR server to find out the location of the caller and then sends the SIP message to caller. Proxy can be stateless or stateful. Stateless proxy refers to one, which does not maintain your call state. Stateful proxy maintains the call state.

REDIRECT - Redirect server works opposite to what proxy does. If SIP server is configured in redirect mode and you send the SIP message to SIP server to call someone, redirect server returns you to the current location of caller in SIP response and then you yourself can send the SIP message to the caller.

Q. What is SIP and SIP clients?

A. SIP is a signaling protocol. SIP client refers to the soft or hard devices, which can register with SIP server for example IP phones, ATA, VoIP gateways, etc.

Q. What is Quality of Service? One QOS is what we see when we run gpedit.msc in our PC. Is that the same or any other application or what?

A. I will answer this question in my next hub.

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4 comments

Veronica Allen profile image

Veronica Allen 7 years ago from Georgia

Good job sonni2006. I appreciate how you broke things down for us technically challanged individuals.


soni2006 profile image

soni2006 7 years ago from New Delhi, India Author

Thanks a lot Veronica. These were some questions asked by someone via the email to which I have provided answers to via the above hub. I am looking forward to making more hubs on technology in the upcoming days, especially IP phones and voip phones functions, advantages, disadvantages, security, use, and benefits.


Graham Allen 5 years ago

You can have issues with SIP behind a symetric NAT. Symetric NAT's are the most common and this link explains why you will expeience one way audio or none at all. http://think-like-a-computer.com/2011/03/14/fix-on...


soni2006 profile image

soni2006 5 years ago from New Delhi, India Author

One way audio is totally useless for a system like VOIP. Thank you so much Graham for sharing this info.

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