VoIP Interview Questions with answers:
What is VoIP?
VoIP stands for Voice
over Internet Protocol. It means the transmission of voice and call control
data over the Internet. In other words, this technology allows you to make
phone calls over the Internet.
How does VoIP work?
To understand how VoIP
works, you will be taken through the process of voice transmission from one end
to the other. The process starts with a person talking into the mouthpiece on
one end of a VoIP call.
This analog voice
signal must first be sampled and digitized. Voice sampling is usually done
8,000 times per second (8KHz). In order to reduce bandwidth, a voice CODEC is
used. A voice CODEC is a compression/decompression algorithm that is optimized
for the voice frequency range. The bit stream uncompressed is 64Kbps. By using
an available CODEC, the bit stream can be reduced to 8Kbps or less.
In order for the
compressed voice data to be sent over the Internet, it must go through a
process called packetization. This is a packet consisting of a small sample of
the voice data (usually 10-30 milliseconds).
While being routed
through the Internet, these packets can get delayed or even lost. This can
cause degradation in voice quality. Simply put, there are various mechanisms in
place to compensate for these problems and help smooth out the audio.
Once all the packets
arrive on the listening end of the call, they must be reassembled to their
original state. The packets are decompressed and converted from a digital to
analog voice signal.
What equipments are needed for VoIP?
Generally following
things are required for voip
- Broadband connection
- voip phone
- nexton softswitches
- router
- audiocodec
- astric server
What are the advantages to VoIP?
The big advantage is
VoIP may save you money depending on how much you are currently spending for
local and long-distance calls. What you will need to do is get the total cost
the phone company is charging and compare it against a VoIP plan that interests
you. With most plans, you get free calls within the U.S. and Canada for a low
flat rate. International calls usually have very low rates with no connection
fees. For both residential customers and businesses that make a lot of long
distance and international calls, the savings can be several hundred dollars a
year.
Another advantage is
with the features available with VoIP. Features such as caller ID, call
waiting, call forwarding, 3 way conferencing and voice mail are usually
included at no extra cost. With the phone company, these services are usually
extra.
In addition, you can
make free phone calls anywhere there is a high speed Internet connection
available. That means you can be in another state or even in another country
and make calls as if you were back at your home or business. You will just need
to bring your phone adapter along with you and possibly a phone in case one is
not available.
What do internet telephony,packet telephony,IP
telephony and converged network means?
The first thing all
mean the same thing. Which is using IP (Internet protocol) for voice services.
Some voice networks are only packet-switched and have no access outside of
their own VoIP network. Most VoIP networks have a Gateway that connects to a
circuit-switched external network which gives them acces to external calling.
One of the gateways responsibilites is to convert G.711 Circuit-switched media
(typically a T1 provided by a telco company) to the 7.723 Packet-switched media
that will traverse the companies VoIP network. A device called a gatekeeper
will then convert the IP address (used by H.323 protocol) to a standard
telephone number (E.164 address) that can be used for external calling.
A converged network is
a network that passes both Voice and Data over the same set of devices. Converged
networks generally implement QoS (Quality of service) on all actived network
devices to ensure the VoIP has priority over standard data because of it's more
rigid demands.
What Are Some disadvantages of VoIP?
If you're considering
replacing your traditional telephone service with VoIP, there are some possible
differences:
- Some VoIP services don't work during power outages and the service provider may not offer backup power.
- Not all VoIP services connect directly to emergency services through 9-1-1.
- VoIP providers may or may not offer directory assistance/white page listings.
How does SIP support
caller ID?
Caller-ID is provided by the From SIP header containing the
caller's name and "number". The number would most likely be placed in
the user field of a SIP URL or appear in a tel: URL.
Since the callee generally does not know or trust the callee's
server, only cryptographic signatures can be used to ensure that the
information is valid. For example, the outgoing proxy might be operated by an
ISP, enterprise or phone company and sign for the identity of the caller, using
the signedby parameter, with the identity of the company verified by a public
key certificate similar to those used by web sites.
What is the difference
between a call leg and a call id?
A call leg refers to the one-to-one signaling relationship between
two user agents (UAs). The Call-ID is an identifier, carried in the SIP
messages, that refers to the call. A call is a collection of call legs. A UAC
starts by sending an INVITE; because of forking, it may receive multiple 200
OKs from different UAs. Each corresponds to a different call leg within the
same call. Call is thus a grouping of call legs. In the call control spec,
additional call legs are created through the Also header.
Call legs refer to end-to-end connections between user agents,
rather than any relationship with proxies. Within a call leg, there are
numerous transactions in both directions.
The request URI is not used in call leg identification.
The To and From field relate to local and remote in the following
way. When Alice sends a request on a call leg to Bob, the From field contains
the local address (Alice), and the To field the remote address (Bob). When a
request is received by Bob, the To field is matched to Bob's local address, and
the From field to the remote address (Alice).
The CSeq spaces in the two directions of a call leg are
independent. Within a single direction, the sequence number is incremented for
each transaction.
What is the difference
between tag and branch-id?
Branch IDs allow proxies to match responses to forked requests. Without
them, a proxy wouldn't be able to tell which branch a response corresponds to.
Tags, in To headers, are of no help here since they are not known until
responses arrive. Tags are used by the UAC to distinguish multiple final
responses from different UAS.
A UAS has no reliable way of determining if the request has been
forked or not. Thus, to be safe it needs to add a tag. Proxies only insert tags
into the final responses they generate themselves; they never insert tags into
requests or responses they forward.
Since a request can be forked several times on its way to UAS, a
single "tag" (or whatever you like to call it) added to the request
by one of the proxies is not sufficient for the next forking proxy along the
chain to match responses on its own branches; every proxy that forked the
request would need to add its own unique IDs to the branches it created. This
is precisely what's being achieved by the branch parameter in the Via header.
(Igor Slepchin)
How can one recognize a
retransmitted, duplicate or looped request?
header retransmitted
duplicate matching response From same same same To same same same, but tag may
have been added Call-ID same same same request same same same URI CSeq same
same same Via - - must be local host; check for branch parameter to identify
which branch
Looped request are
recognized by one or more of the following:
·
The server finds itself
in the request's Via list, including any
branch parameter. (The server should compute the
branch parameter so that it depends on the request URI.)
·
The server is about to
proxy the request to one of the hosts listed
in the Via list. The same
·
The Max-Forward count is
decremented to zero.
·
The Expires time has
elapsed.
What is the relationship
between the From, Contact, Via and Record-Route/Route headers?
All these headers determine how requests and responses are routed
in a network of SIP proxy servers. Roughly, the distinction is:
From:
Used for subsequent
requests if there is no Contact or Record-Route header. E.g., if Alice makes a
call with From: Alice <alice@example.org> to Bob, an INVITE request from
Bob to Alice would use alice@example.org as the To header and Request-URI.
Contact:
Determines the
destination placed in the Request-URI for subsequent requests and can be used
to bypass proxies not enumerated in a Record-Route header. Also used in
responses by redirect servers and in REGISTER requests and responses.
Record-Route/Route:
The Record-Route header
is inserted into requests by proxies that want to be in the path of subsequent
requests for the same call-id. It is then used by the user agent to route
subsequent requests. The mechanism is similar to a source-route, copying the
Record-Route information into a set of Route headers. The Request-URI is set to
the first Route header.
Via:
Via headers are inserted by servers into requests to detect loops
and to allow responses to find their way back to the client. They have no
influence on the routing of future requests (or responses).
Generally, in short, requests should be sent to Route if present,
Contact if there is no Route, From if there is no Contact.
What's the difference
between the request URIs tel:+12125551212 and sip:12125551212@gw.com?
Non-SIP URLs, such as tel:+12125551212 for a telephone number, may
be used as request URIs in SIP INVITE requests. This only makes sense if all
outbound calls are handled by a proxy server. In the case of a tel: URL, the
proxy server would then translate the request URL to a SIP URL of a gateway
server, if it is not handling the gateway duty itself. The proxy server might
use the Gateway Location Protocol (GLP) to find the appropriate next-hop SIP
server. The To header may always be a tel: URL even if the Request-URI is a SIP
URL, although that breaks with the common practice that Request-URI and To start
out the same.
Is the domain of the
request-URI and the To header always the same?
The Request-URI names the destination of the registration request,
i.e., the domain of the registrar. The user name must be empty. Generally, the
domains in the Request-URI and the To header field have the same value;
however, it is possible to register as a "visitor", while maintaining
one's name. For example, a traveler sip:alice@acme.com (To) might register
under the Request-URI sip:atlanta.hiayh.org, with the former as the To header
field and the latter as the Request-URI. Note, however, that requests for a
user at acme.com are not likely to arrive at the atlanta.hiayah.org server;
special purpose routing logic will generally need to be established in order
for requests for alice@acme.com to go to the atlanta.hiayh.org server . In the
vast majority of cases, the domains in the request URI and To field will match.
The REGISTER request is no longer forwarded once it has reached the server
whose authoritative domain is the one listed in the Request-URI.
How do I ensure
registrar reliability?
There are several techniques that can be used to minimize the
impact of registrar/proxy server failures for a server in a local area network:
- Run several servers that all respond to the same multicast registration
address ("warm standby"). As long as multicast requests
are mostly reliable, this ensures a consistent registration picture.
- If a registration server is rebooted and does not have complete knowledge
of the local UA population, it could multicast any incoming INVITE
requests.
For servers separated from their client by a wide-area network,
use of multicast is not appropriate, so that these servers have to rely on
traditional backup techniques to achieve reliability. For example, the
designated registrar could multicast registration updates within its local
network to keep standby servers synchronized.
Are ACK requests
retransmitted?
No. An ACK is sent when a response retransmission is received.
Reliability is achieved because the response is retransmitted until an ACK
arrives, and the ACK is retransmitted on response retransmissions. ACK is only
used for INVITE.
How are BYE requests
routed?
Since a 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 a Record-Route, it will traverse the list of proxies indicated there.
If the caller decides to send a BYE before receiving a 200 from
the callee, the BYE is be handled by the proxies just as the corresponding
INVITE was handled, i.e., it may be forked.
Can I CANCEL requests
other than the first INVITE?
Yes, any request can be cancelled before it has been executed by
the UAS. However, it is likely that this will only make sense in practice for
the initial INVITE and subsequent "re"INVITE. In the latter case, the
call remains, just any changes requests are cancelled.
How does a caller find
its proxy server?
Calls typically proceed directly to the callee's domain. For
example, when calling alice@example.com, the INVITE request would be sent to
the SIP server for the domain example.com, found via DNS.
If a "local" (outbound) proxy is needed for outgoing
calls, it currently needs to be manually configured, similar to the
configuration of web proxies in browsers. Extensions to (for example) use a
REGISTER response or DHCP are under discussion.
What's the difference
between a stateless and a stateful proxy server?
Stateless proxies forget about the SIP request once it has been
forwarded. Stateful proxies remember the request after it has been forwarded,
so they can associate the response with some internal state. In other words,
stateful proxies maintain transaction state. Stateful implies transaction
state, not call state.
Stateless proxies scale very well, and can be very fast. They are
good for network cores. Stateful proxies can do more (they can fork, for
example, see the next question) and can provide services stateless ones can't
(call forward busy, for example). They don't scale as much as stateless ones. An
admininstrator gets to decide which to use. These are also logical entities; a
physical proxy is likely to act as a stateless proxy for some calls, stateful
for others, and as a redirect server for even others.
Neither stateful nor stateless proxies need to maintain call
state, although they can, but will need to make sure that they are part of
subsequent transactions via the Record-Route header.
Proxies must be stateful if one of the following conditions hold:
- uses TCP,
2. uses multicast, 3. forks.
What is the
datastructure used to save the call-ids.
What is the use of 183
in sip, and where it is used.
1. What is the
difference between a dialog and a transaction ?
2. Do I always need to
use a proxy server/ why is a proxy server required?
3. What is the difference
between tag and branch-id?
4. Why can a forking SIP
proxy not be stateless?
5> how do you
determine if the REGISTER message sent from a client is a Re-Registration ?
6> What is the
difference between the definitions of a REGISTER and a DEREGISTER message ?
7> Is there any
difference on how the REQUEST-URI is framed for a REGISTER message w.r.t an
INVITE ?
If yes what is it ?
1. Why CANCEL is not be
challenged?
2. What is the
difference between INVITE and re-INVITE?
3. What is 'command
sequence' is SIP? What is the use of that?
4. What is the
difference between Dialog and Transaction?
5. It is said that ACK
is seperate transaction? But why the C-Seq is same with the INVITE?
6. What is the use of
VIA, contact header, TO header, from header, Route headedr? How are they
different?
7. What is Strict
routing and loose routing in SIP?
8. Have u heard about
Magic cookie? What is that and what is that implyes?
9. How many mandatory
headers are there in a SIP request? What are they?
10. What is the use of
ACK message in INVITE transaction? If TCP is used as transport then can we
ignore sending of ACK?
11. What is offer-answer
model?
12. What is the
difference between INVITE transaction and non-Invite Transaction?
13. Can you send CANCEL
immidiately after sending an INVITE msg? Then when will it be sent? Does
sending of CANCEL ensure canceling of INVITE session? If not what needs to be
done?
14. What is B2BUA
functionality and what is Proxy functionality in SIP?
15. How do u
differentiate a refresh registration message and a de-registratoin mesage?
16. A user can register
himself from different location simultaneously. So how can u get all the
bindings of an user from the server?
17. What is parallel
forking and sequencial forking? In case of Parallel forking how an incoming
responses from different branches are identified at proxy?
19. Have you heard about
Presence? what is that?
20. Extract CallID
header with it's value from a buffer contains SIP message only. No SIP message
parser available for this.
1: What is diff between
invite and reinvite. 2: Which field is used for displaying call id. 3: What is
SDP. Is there rules for the order of SDP field. 4: What is privacy header. 5:
What is passerted header. 6: What is session level and media level attributes
in SDP.
1)Wat is a Sip
Transaction.
2)How do u differentiate
bn Reister and DeRegister
3)How should a registrar
behave if CSeq of refresh register is less than previous register
4)Wat is third party
registration.
5)how r the from and to
headers in third party registration.
6)Explain Invite
transaction
7)Invite Success case,
Invite Failed case.
8)authentication for
ACK, CANCEL.
9)Explain the process to
cancel an ongoing transaction. PreRequisties,
10) Wat is a sip dialog
11) which messages
initiaite a dialog
12) what is difference
between invite and reinvite.
13) explain the process
to setup a sip session, and dismantling a session
14) what is the
behaviour of sip proxy in case callee do not send 200 ok for invite
15) what is the sip
proxy behaviour in case callee is not registered
16) significance of
route header
17) record header
18) via header
19) how do u handle
transactions, (key used) to check implemetation knowledge
20) significace of to,
from callid, cseq
21) what is dialog id
22) is invite indialog
request?
23) explain header
modifications in setting up a dialog
1. What is the difference between a dialog and a transaction ?
2. Do I always need to use a proxy server/ why is a proxy server required?
3. What is the difference between tag and branch-id?
4. Why can a forking SIP proxy not be stateless?
5. how do you determine if the REGISTER message sent from a client is a Re-Registration ?
6. What is the difference between the definitions of a REGISTER and a DEREGISTER message ?
7. Is there any difference on how the REQUEST-URI is framed for a REGISTER message w.r.t an INVITE ? If yes what is it ?
1. Why CANCEL is not be challenged?
2. What is the difference between INVITE and re-INVITE?
3. What is 'command sequence' is SIP? What is the use of that?
4. What is the difference between Dialog and Transaction?
5. It is said that ACK is seperate transaction? But why the C-Seq is same with the INVITE?
6. What is the use of VIA, contact header, TO header, from header, Route headedr? How are they different?
7. What is Strict routing and loose routing in SIP?
8. Have u heard about Magic cookie? What is that and what is that implyes?
9. How many mandatory headers are there in a SIP request? What are they?
10. What is the use of ACK message in INVITE transaction? If TCP is used as transport then can we ignore sending of ACK?
11. What is offer-answer model?
12. What is the difference between INVITE transaction and non-Invite Transaction?
13. Can you send CANCEL immidiately after sending an INVITE msg? Then when will it be sent? Does sending of CANCEL ensure canceling of INVITE session? If not what needs to be done?
14. What is B2BUA functionality and what is Proxy functionality in SIP?
15. How do u differentiate a refresh registration message and a de-registratoin mesage?
16. A user can register himself from different location simultaneously. So how can u get all the bindings of an user from the server?
17. What is parallel forking and sequencial forking? In case of Parallel forking how an incoming responses from different branches are identified at proxy?
19. Have you heard about Presence? what is that?
20. Extract CallID header with it's value from a buffer contains SIP message only. No SIP message parser available for this.
21. what is path MTU ?
2. Do I always need to use a proxy server/ why is a proxy server required?
3. What is the difference between tag and branch-id?
4. Why can a forking SIP proxy not be stateless?
5. how do you determine if the REGISTER message sent from a client is a Re-Registration ?
6. What is the difference between the definitions of a REGISTER and a DEREGISTER message ?
7. Is there any difference on how the REQUEST-URI is framed for a REGISTER message w.r.t an INVITE ? If yes what is it ?
1. Why CANCEL is not be challenged?
2. What is the difference between INVITE and re-INVITE?
3. What is 'command sequence' is SIP? What is the use of that?
4. What is the difference between Dialog and Transaction?
5. It is said that ACK is seperate transaction? But why the C-Seq is same with the INVITE?
6. What is the use of VIA, contact header, TO header, from header, Route headedr? How are they different?
7. What is Strict routing and loose routing in SIP?
8. Have u heard about Magic cookie? What is that and what is that implyes?
9. How many mandatory headers are there in a SIP request? What are they?
10. What is the use of ACK message in INVITE transaction? If TCP is used as transport then can we ignore sending of ACK?
11. What is offer-answer model?
12. What is the difference between INVITE transaction and non-Invite Transaction?
13. Can you send CANCEL immidiately after sending an INVITE msg? Then when will it be sent? Does sending of CANCEL ensure canceling of INVITE session? If not what needs to be done?
14. What is B2BUA functionality and what is Proxy functionality in SIP?
15. How do u differentiate a refresh registration message and a de-registratoin mesage?
16. A user can register himself from different location simultaneously. So how can u get all the bindings of an user from the server?
17. What is parallel forking and sequencial forking? In case of Parallel forking how an incoming responses from different branches are identified at proxy?
19. Have you heard about Presence? what is that?
20. Extract CallID header with it's value from a buffer contains SIP message only. No SIP message parser available for this.
21. what is path MTU ?
1. What is VOIP? What is
the Common problem you will face while using Voip Service?
2. What is SIP? What is the major discovery in SIP if compare with Other Protocols?
3. What is SIP Protocol Structure?
4. What are L2 and L3 Layers?
5. What are the general headers you will see in SIP Requests and Responses?
6. How to do SIP conformance Testing?
7. What kind of scripts you know to automate SIP Call flow?
8. How to check usability testing on SIP Phone?
9. How you will do Compatibility testing on Voip?
10. Give me an example when servers send 5XX response?
2. What is SIP? What is the major discovery in SIP if compare with Other Protocols?
3. What is SIP Protocol Structure?
4. What are L2 and L3 Layers?
5. What are the general headers you will see in SIP Requests and Responses?
6. How to do SIP conformance Testing?
7. What kind of scripts you know to automate SIP Call flow?
8. How to check usability testing on SIP Phone?
9. How you will do Compatibility testing on Voip?
10. Give me an example when servers send 5XX response?
