Here we present some SIP Interview Questions& Answer:

1)  what is Media Negosiation ?If SDP is not sent in Invite,when that will be sent?

A: Media Negotiation is only trade of Media parameters needed to secure the session.There is a two- stage trade done in Invite and 200 OK ,transaction abilities is in view of essential Offer/Answer model of SDP exchnage.
Note: If SDP is not sent in Invite,  then it can sent in ACK ask. 
  

2) What is dialog? How would we distinguish a dialog?

An: A beginning appeal from UAC contains a tag in  "From" header and Call ID. As of right now we have half dialog finished. What's more, tag in "To" header is included by UAS in Provisional reactions other than 100-Trying.This finishes the "dialog"
Dialog = From label + To label + Call Id 

3) What is Transaction? The most effective method to recognize?

An: A Request took after by the Final Response is known as an "Exchange".
It is recognized by "C-seq" and "Branch Parameter". 

4) What is Session? At the point when does Session is Established?

A:  Session is  trade of media between two or more endpoints. In the wake of receving ACK ask for just Session is established.If we don't get ACK ,session is not Established. 

5)  If Max-Forwards reaches to Zero, what reaction is sent?

A:  483-Too Manys Hops reaction is sent from intermediary. 

6) How would we be able to perceive a retransmitted, copy or circled solicitation?

A:  a) The maximum forward check is decremented to zero.
b) The Expires time has slipped by.
c) The Server ends up in the solicitations by means of rundown, including any branch parameter.

7) Can you please portray SDP ?

A: SDP Describes
Session name and reason
Time(s) the session is dynamic.
The media involving the session
Data expected to get those media (locations, ports, groups, and so on.)
Data about the transfer speed to be utilized by the session
Contact data for the individual in charge of the session

8) How would we be able to realize that call is on Hold?

A:  1) If  SDP contains a characteristic a=sendonly or Inactive, then call is on hold.
2) Zeroing the IP address or port number in the media descriptor "c"of the stream.

11)  What are compulsory header fields?

A :  To, From, Via, C-seq, Max Forwards,  Call-ID.

12)  What is Forking Feature and forking sorts?

A:  It sends an Invite appeals to all the accessible clients.
Two Types of Forking.
Serial Forking:  In this  it sends appeal to one location, if  that fizzles then it attempt second  address.
Parallel Forking : In this it sends demand at once to all addresses,If any of the client  acknowledge the solicitation others get detached.

13)  What is Call-ID ?

A:  It contains a universally one of a kind identifier for all solicitations and reactions sent by either UA in a  dialog, and it is produced by the mix of 'arbitrary string' and 'IPaddress'.

14)  If client a doesn't have upheld media what reaction is sent?

An: UnSupported Media -415 is sent from intermediary if Codecs doesn't coordinate .

15)  what are SIP Entities/Components?

A:  There are four Types :
a) Registrar Sever
b) Proxy Server
c) Redirect Server
d) UserAgent Server

16)  What are 1xx-responess do you know?

A:
Educational =  "100" ; Trying
|  "180"  ;  Ringing
|  "181"  ;  Call Is Being Forwarded
|  "182"  ;  Queued

17)  What is enchantment treat?

A:  The branch ID embedded by a component dependably start with the characters "z9hG4bK".These 7 characters are utilized as an "Enchantment treat"

19) Does SIP convey DTMF?

A:  There are atleast two alternatives for conveying DTMF and smilar motions in a Voip N/W utilizing  SIP.First DTMF can be transported as a RTP payload. This has the preference that it  gives precise timing  and alingment with RTP bundle as of now there is no institutionalized  arrangement with in SIP, yet it has been proposed to convey DTMF data in SIP Info messages.

20) Do guest need to know the area of the area server?

A:  The guest doesn't cooperate straightforwardly with the area server. A sidetrack or intermediary server approaches  the area  serve for guidance.

21) Does SIP do affirmation control and oversee Band width?

A:  1) Since these offers no genuine security affirmation control is not bolstered by SIP.
2) No, that is the part of an asset reservation convention.

22) Do i generally require an intermediary server or sidetrack server?

A:  Proxy and Re-direct server are intelligent Entities.So,sip servers can get in touch with one another  straightforwardly.

23) How does guest think that its neighborhood enlistment center?

A:  The neighborhood enlistment center is physically designed.

24) Are Ack appeals retransmitted?

A:  No, An Ack is sent when a reaction retransmission is gotten. Ack is utilized for Invite.

25) How are BYE appeals steered?

A:  Since the contact header must be introduce in Invite and 200. The BYE will go straightforwardly to the  client specialists if there is no record-course header. On the off chance that there is record-course it will cross the rundown of  intermediaries demonstrated here.

26) Can I cross out  an appeal other than the first Invite?

A:  Yes, any solicitation can be can scratched off before it has be executed by UAS.

27) How does a guest think that its intermediary server?

A:  Calls normally continue specifically to the callee area.

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

A:  A forking SIP intermediary can't be stateless on the grounds that it needs to perform a separating operation,  giving back one reaction out of numerous it gets.

29) Does SIP do keep alive?

A:  SIP itself doesn't have a keep-alive component amid the call.

30) What is connection in the middle of MGCP and SIP?

A:  MGCP is utilized in the middle of MG and MGC. Taste may be utilized between two controller for companion to  peer association. just MGC needs to comprehend both conventions.

31) Can H.323 and SIP utilized together?

A:  Yes, that permits SIP and H.323 terminals  to call one another.

32) How would I interconnect ISUP and SIP?

A:  SIP can be utilized between SS7 hubs. while all points of interest have not been worked out, the essential  call stream is like ISDN case.

33)  What is the contrast between Transaction, Dialog and Session?

A:  REFER Q2 & Q3 & Q4.

34) How circle can be recognized ?

A :  Looped solicitation can be perceived in taking after way:
The Max-Forward checks is decremented to zero.
The Expires time has slipped by.
The server ends up in demand's VIA rundown including any branch parameter.

35) What is Call stream of meeting ?

36) What is Call stream of two gathering session?

37) If An accept feature call and B accept sound call , how A comes to realize that be is accepting Audio call ?

A:

38) What is SDP ? In what manner would we be able to know there is an Audio/ feature call?

An: A SDP session portrayal incorporates the accompanying media data:
o  The sort of media (feature, sound, and so on.)
o  The vehicle convention (RTP/UDP/IP, H.320, and so on.)
o  The arrangement of the media (H.261 feature, MPEG video,G.711 Audio,G.729 Audio and so on.)
The SDP Parameter  "m=Media Type" portrays it is an Audio/Video Call.

39) If Max-advances scopes to zero.then what happens?

41) What is an Early Dialog?

A:  A dialog created by a non-last reaction to an appeal  is in "Ahead of schedule " state and it is additionally called "Early Dialog".

42) what is SIP URI?

43) What is VOIP?

A: VOIP characterizes Voice Over Internet Protocol. It is the web innovation to convey  voice correspondence and mixed media sessions over Internet convention systems, for example, web.

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

A: SIP(Session Initiation Protocol) is a Signaling  Protocol. It is utilized to Initiate,Modify and Terminate session. It lies  on Application Layer. If you don't mind allude beneath outline.















45) What are SIP Components?

A:  There are two taste segments :-
a) User Agent (UA)
b) SIP Servers.

46) What are Sip Servers?

A:  Sip servers are taking after sorts
Intermediary Servers
Enlistment center Servers
Divert Servers

47)  What are Codecs?

A: Codec (pressure decompression) is a calculation which packs and decompresses a voice packet.The G.711 codec to change over a simple voice to a digitized voice stream.The most generally utilized codec as a part of VoIP environment is G.729.

48) How would you offer IP to Endpoint?

A:  Manually  and Using  DHCP.

49) In DHCP who will relegate IP to endpoint?

A: DHCP Server will relegate the  IP location to the Endpoint.

50) Explain how IP location is alloted to endpoint utilizing DHCP convention?

A:  The Dynamic Host Configuration Protocol (DHCP) is a system convention used to arrange gadgets that are joined with a system so they can convey on that system utilizing the Internet Protocol (IP).
DHCP operations fall into four fundamental stages: IP disclosure, IP lease offer, IP solicitation, and IP lease affirmation.

















51) Explain how would you enlist Endpoint  and where do you enroll?

A: Please allude the connection underneath:
http://siptestingknowledge.blogspot.in/p/elements call-streams 1-enlistment center server.html
We enlist client with Register Server .

52) What are obligatory Header  Fields in register demand?

A: To ,From,Call Id, C-seq, lapses...

53) Explain Call stream of register?

A: Please allude the connection beneath:
http://siptestingknowledge.blogspot.in/p/substances call-streams 1-recorder server.html

54) In which message www- confirm/Authorization header fields utilized ?

A:  401-Unauthorized reaction contains header field "WWW-Authenticate", where Register appeal contains header field "Approval".

55) In which message Proxy-Authenticate/Proxy-Authorization header fields are utilized?

A: 407-Proxy Authentication obliged reaction contains header documented  "Intermediary Authenticate" and Invite appeal contains "Intermediary Authorization".

56) Does Initial Register solicitation contains Authorization header field?

A:  No.

57) Explain configuration of Request URI, TO, From in Register Request?

A: Request URI:- taste: atlanta.com
From:- taste: UserA@atlanta.com
To:- taste: UserA@atlanta.com

58) What responce is gotten if area wrong while client register?

A:  500 -Cannot focus domain.

59) What responce is gotten when client dial Wrong number?

A:  404-Not Found reaction is gotten.

60) What is Tag parameter?

A:  Tags are utilized by the UAC to recognize various last reactions from diverse  UAS.
A starting appeal from a customer will contain a From Tag and the consequent temporary reaction to it from the server will contain a To Tag.

62) What are the sorts to keep approach hold? Have you tried keeping  ip adderss to 0.0.0.0 with port number accessible?

A:  1) If  SDP contains a property a=sendonly or Inactive, then call is on hold.
2) Zeroing the IP address or port number in the media descriptor "c"of the stream.

63) What are supplementary features(Call highlights)?

A: Call forward,Transfer,Conference,Busy ,Hold....etc

64) Explain call stream about Transfer Unattended?

A: Please allude Link http://siptestingknowledge.blogspot.in/p/substances call-streams 1-enlistment center server.html.

66) Do we get Refer –by header in allude ask for in exchange ? Why it is utilized , if as of now User A knows User B?

A:  Yes, allude By is a discretionary header field,mostly it contains in exchange .

67) Why 202 -Accepted is sent for Refer, rather than 200-Ok in Transfer unattended?

A: 200 Ok is last reaction,

68) What header fields does Notify contains sent from User B to User An in Transfer unattended?(Regarding  attempting to C)

A:  It contains two header fields:
Every NOTIFY MUST contain an Event header field with an estimation of allude
Occasion = allude
Membership state = active,expires=60
Every NOTIFY MUST contain a group of sort "message/sipfrag"
Substance sort = message/sipfrag
SDP conatins " Sip2.0 100 Trying "

69) If as of now User A distinction from bring in exchange Un went to, why User B will Notify  the  User A  after call accomplishment with User C?

A:  User A doesn't detach from call  regardless of the fact that sent BYE
demand , the dialog between  User An and  B still exists until the Subscription made by the REFER has ended.
Advise appeal sent by User B to A contains the  header fields
Occasion = allude
Membership state= terminated,reason= noresource

70) what happens if REFER appeal contains more than one allude To field Values?

A:  An operators reacting to a REFER strategy MUST give back a 400 (Bad Request)
on the off chance that the solicitation contained zero or more than one Refer-To header field
values.

71) How to identify numerous REFER asks for in a dialog?

A: This id parameter MAY be incorporated in NOTIFYs to the first REFER a UA gets in a given dialog. A SUBSCRIBE sent to revive or end this membership MUST contain this id parameter.
Occasion: refer;id=93809824
The number from the CSeq header field estimation of REFER is given as id number in Notify.
EX:
Allude sip:b@atlanta.example.com SIP/2.0
Through: SIP/2.0/UDP agenta.atlanta.example.com;branch=z9hG4bK9390399231
To: <sip:b@atlanta.example.com>;tag=4992881234
From: <sip:a@atlanta.example.com>;tag=193402342
Call-ID: 898234234@agenta.atlanta.example.com
CSeq: 93809824 REFER
Max-Forwards: 70
Allude To: (some distinctive URI)
Contact: sip:a@atlanta.example.com
Content-Length: 0
Inform sip:a@atlanta.example.com SIP/2.0
Through: SIP/2.0/UDP agentb.atlanta.example.com;branch=z9hG4bK9320394238995
To: <sip:a@atlanta.example.com>;tag=193402342
From: <sip:b@atlanta.example.com>;tag=4992881234
Call-ID: 898234234@agenta.atlanta.example.com
CSeq: 1993404 NOTIFY
Max-Forwards: 70
Occasion: refer;id=93809824
Membership State: active;expires=(depends on Refer-To URI)
Contact: sip:b@atlanta.example.com
Substance Type: message/sipfrag;version=2.0
Content-Length: 20
Taste/2.0 100 Trying

73) What are Voip Components ?

A: There are different segments which means make voip successfull. Taking after are the voip parts:
IP Phones
Doors
Watchmen
MCUs
Application Server
Call Agent
IP Phones
Handset, or simple telephone joined with a voip connector.
Doors
The door gives interpretation in the middle of VoIP and non-VoIP systems, for example, the PSTN.
Watchmen
Gives CAC (Call Admission Control) or and transmission capacity administration. Call affirmation control is a methodology used to guarantee, or keep up, a certain level of sound quality in voice correspondences systems, or a certain level of execution in Internet hubs and servers.
MCUs
Multipoint Control Units gives the usefulness of call, feature conferencing.
Application Server
Application server give additional usefulness, for example, phone message, informing and so forth.
Call Agent
(Otherwise called delicate switch or Media Gateway Controller) The Call Agent/Softswitch/MGC gets flagging data (like dialed digits) from the Media Gateway and can educate it to caution the called party, to send and get voice information and so on.
VoIP Functions
Like customary telephony, VoIP obliges some capacity through which a call can be finished. Taking after are the capacities:
Flagging
Flagging is the capacity to create and trade call control data that will be utilized to secure, screen, and discharge associations between two endpoints. PSTN Network utilizes SS7 (out of band) as a vehicle to trade messages, on the other hand. VoIP system utilizes H.323, SIP, MGCP, SCCP as flagging. These join and disengage messages are completed by SS7 if there should arise an occurrence of PSTN. In VoIP environment these messages are completed by SIp and H323. Taste and H.323 are shared flagging conventions where the end gadgets or entryways contain the brainpower to start and end call sessions and translate call control messages.
Database Services
Database administrations incorporate access to charging data, guest name conveyance (CNAM) and so on. CNAM is a keen administration which shows the guests name in the calling party's telephone rather than the guest ID. Database benefits additionally incorporate access to calling cards. Another illustration of Database Service is giving a call notice administration which puts outbound calls with prerecorded messages at particular times to inform clients of occasions like new plans and bundles, wake-up calls and so on.
74) Can you please clarify the callflow of Two-gathering session?


Two-gathering Call

For two-gathering Internet telephone calls, the appeal must contain a

depiction of where to send the information. In the illustration above, Bell

calls Watson. Ringer shows that he can get RTP sound codings 0

(PCMU), 3 (GSM), 4 (G.723) and 5 (DVI4).

C->S: INVITE sip:watson@boston.bell-tel.com SIP/2.0

By means of: SIP/2.0/UDP kton.bell-tel.com

From: A. Ringer <sip:a.g.bell@bell-tel.com>

To: T. Watson <sip:watson@bell-tel.com>

Call-ID: 3298420296@kton.bell-tel.com

CSeq: 1 INVITE

Subject: Mr. Watson, come here.

Substance Type: application/sdp

Content-Length: ...

v=0

o=bell 53655765 2353687637 IN IP4 128.3.4.5

s=Mr. Watson, come here.

c=IN IP4 kton.bell-tel.com

m=audio 3456 RTP/AVP 0 3 4 5

S->C: SIP/2.0 100 Trying

By means of: SIP/2.0/UDP kton.bell-tel.com

From: A. Ringer <sip:a.g.bell@bell-tel.com>

To: T. Watson <sip:watson@bell-tel.com> ;tag=37462311

Call-ID: 3298420296@kton.bell-tel.com

CSeq: 1 INVITE

Content-Length: 0

the gathering of the call is affirmed promptly with(100)

S->C: SIP/2.0 180 Ringing

By means of: SIP/2.0/UDP kton.bell-tel.com

From: A. Ringer <sip:a.g.bell@bell-tel.com>

To: T. Watson <sip:watson@bell-tel.com> ;tag=37462311

Call-ID: 3298420296@kton.bell-tel.com

CSeq: 1 INVITE

Content-Length: 0

perhaps after some database mapping defer, the call rings (180)

S->C: SIP/2.0 200 OK

By means of: SIP/2.0/UDP kton.bell-tel.com

From: A. Ringer <sip:a.g.bell@bell-tel.com>

To: <sip:watson@bell-tel.com> ;tag=37462311

Call-ID: 3298420296@kton.bell-tel.com

CSeq: 1 INVITE

Contact: sip:watson@boston.bell-tel.com

Substance Type: application/sdp

Content-Length: ...

v=0

o=watson 4858949 IN IP4 192.1.2.3

s=I'm on my way

c=IN IP4 boston.bell-tel.com

m=audio 5004 RTP/AVP 0 3

Watson can just get PCMU and GSM. Note that Watson's rundown of

codecs might be a subset of the one offered by Bell,

Watson  send sound information to port 3456 at c.bell-tel.com

Chime will get them on port 3457.

Chime will send to port 5004 at boston.bell-tel.com.

Watson will get RTCP bundles on port 5005.

Since the two sides have concurred on the arrangement of media, Bell affirms

the call without encasing another session portrayal:

C->S: ACK sip:watson@boston.bell-tel.com SIP/2.0

Through: SIP/2.0/UDP kton.bell-tel.com

From: A. Chime <sip:a.g.bell@bell-tel.com>

To: T. Watson <sip:watson@bell-tel.com> ;tag=37462311

Call-ID: 3298420296@kton.bell-tel.com

CSeq: 1 ACK

Ending a Call

To end a call, guest or callee can send a BYE ask:

C->S: BYE sip:watson@boston.bell-tel.com SIP/2.0

Through: SIP/2.0/UDP kton.bell-tel.com

From: A. Chime <sip:a.g.bell@bell-tel.com>

To: T. A. Watson <sip:watson@bell-tel.com> ;tag=37462311

Call-ID: 3298420296@kton.bell-tel.com

CSeq: 2 BYE

In the event that the callee needs to prematurely end the call, it essentially turns around the To and

From fields. Note that it is improbable that a BYE from the callee will

navigate the same intermediaries as the first INVITE.