SIP Interview Questions 3

25) How are BYE requests routed?

 A:  Since the contact header must be present in Invite and 200.
 The BYE will go directly to the  user agent if there is no record-route header.
If there is record-route it will traverse the list of  proxies indicated here.

26) Can I cancel  a request other than the first Invite?

 A:  Yes, any request can be can cancelled before it has be executed by UAS.

27) How does a caller find its proxy server?

 A:  Calls typically proceed directly to the callee domain.

28) Why can a forking SIP proxy not be stateless?

 A:  A forking SIP proxy cannot be stateless because it needs to perform a filtering operation,  returning one response out of many it receives.

29) Does SIP do keep alive?

 A:  SIP itself doesn't have a keep-alive mechanism during the call.

30) What is relation between MGCP and SIP?

 A:  MGCP is used between MG and MGC. SIP may be used between two controller for peer to  peer connection. only MGC needs to understand both protocols.

31) Can H.323 and SIP used together?

 A:  Yes, There is only one product (Lucent packet star IP) that allows SIP and H.323 terminals  to call eachother.

32) How do I interconnect ISUP and SIP?

 A:  SIP can be used between SS7 nodes. while all details have not been worked out, the basic  call flow is similar to ISDN case.

* 24)  What is the difference between Transaction, Dialog and Session?
A:  REFER Q2 & Q3 & Q4.

26) How loop can be detected ?
A :  Looped request can be recognized in following way:

•   The Max-Forward counts is decremented to zero.
•   The Expires time has elapsed.
•  The server finds itself in request's VIA list including any branch parameter.

27) What is Call flow of conference ?

28) What is Call flow of two party session?

30) If A take video call and B take audio call , how A comes to know that be is taking Audio call ?

A: 

31) What is SDP ? How can we know there is an Audio/ video call?

A: An SDP session description includes the following media information:

   o  The type of media (video, audio, etc.)

   o  The transport protocol (RTP/UDP/IP, H.320, etc.)

     o  The format of the media (H.261 video, MPEG video,G.711 Audio,

G.729 Audio etc.)

The SDP Parameter  "m=Media Type" describes it is an Audio/Video Call.

32) If Max-forwards reaches to zero.then what happens? 

33) What is the difference between Route and Record-Route?
A: 
 Route: The header field is used to force routing for a request through
 the listed set of poxies.
Record-route: The header field is inserted by proxies(B2BUA)
 in a request to force future requests in a dialog to be routed  through
 the proxy.

34) What is an Early Dialog?
A:  A dialog established by a non-final response to a request  is in
"Early " state and it is also called "Early Dialog".

35) what is SIP URI?
 It contains display Name and domain

36) What is VOIP?
A: VOIP defines Voice Over Internet Protocol. It is the internet
 technology to carry  voice communication and multimedia
sessions over Internet protocol networks such as internet.

37) what is SIP? Where does it lies on OSI Layer?

A: SIP(Session Initiation Protocol) is a Signalling  Protocol.
 It is used to Initiate,Modify and Terminate session. It lies
  on Application Layer. Please refer below diagram.

38) What are SIP Components?
A:  There are two sip components :-
                        a) User Agent (UA)
                        b) SIP Servers.

39) What are Sip Servers?
A:  Sip servers are following types

• Proxy Servers
• Registrar Servers
• Redirect Servers

40)  What are Codecs?

A: Codec (compression-decompression) is an algorithm which compresses and decompresses a voice packet.The G.711 codec to convert an analog voice to a digitized voice stream.The most widely used codec in VoIP environment is G.729.

41) What are Voip Components ?

A: There are various components which adds up to make voip successfull. Following are the voip components:

1. IP Phones
2. Gateways
3. Gatekeepers
4. MCUs
5. Application Server
6. Call Agent

IP Phones
Handset, or analog phone connected to a voip adapter.

Gateways
The gateway provides translation between VoIP and non-VoIP networks, such as the PSTN.

Gatekeepers
Provides CAC (Call Admission Control) or and bandwidth management. Call admission control is a process used to ensure, or maintain, a certain level of audio quality in voice communications networks, or a certain level of performance in Internet nodes and servers.

MCUs
Multipoint Control Units provides the functionality of call, video conferencing.

Application Server
Application server provide extra functionality such as voicemail, messaging etc.

Call Agent
(Also known as soft switch or Media Gateway Controller) The Call Agent/Softswitch/MGC receives signalling information (like dialed digits) from the Media Gateway and can instruct it to alert the called party, to send and receive voice data etc.

VoIP Functions
Like traditional telephony, VoIP requires some function through which a call can be completed. Following are the functions:

Signalling
Signaling is the capability to generate and exchange call control information that will be used to establish, monitor, and release connections between two endpoints. PSTN Network uses SS7 (out of band) as a transport to exchange messages, however. VoIP network uses H.323, SIP, MGCP, SCCP as signalling. These connect and disconnect messages are carried out by SS7 in case of PSTN. In VoIP environment these messages are carried out by SIp and H323. SIP and H.323 are peer-to-peer signaling protocols where the end devices or gateways contain the intelligence to initiate and terminate call sessions and interpret call control messages.

Database Services
Database services include access to billing information, caller name delivery (CNAM) etc. CNAM is an intelligent service which displays the callers name in the calling party’s phone instead of the caller ID. Database services also include access to calling cards. Another example of Database Service is providing a call notification service which places outbound calls with prerecorded messages at specific times to notify users of events like new plans and packages, wake-up calls etc.