Thursday, July 14, 2011

VoIP Components

In the VoIP network there must be a functionality similar to the one of the PSTN Network. To connect the VoIP network to the PSTN there is the need of a Hardware called Gateway.

The components of a VoIP Network usually are:
1.IP PBX

2.END USER DEVICES VOIP

3.GATEWAYS/GATEKEEPERS

4.IP NETWORK

IP PBX

It is the call processing server. It terminates all VoIP calls on the PSTN Network. It is usually software based.

END USER DEVICES

They usually are: VoIP phones They use the TCP/IP protocol to communicate with the IP network, they have an IP address for the subnet on which they are installed. Usually, VoIP phones use DHCP to auto-configure themselves, with the DHCP server telling the phone about the location of the configuration server, which most of the time is identical to the call processing server.

Softphones

They are software application running on computers, They are usually installed in mobile devices. They have the same base features as VoIP phones.

VOIP GATEWAYS/GATEKEEPERS

Sometimes you can use indifferently the name Gateway or Gatekeeper. The last usually performs the Call admission and bandwidth control. But it is also possible to find all functions in one only hardware. The main function is to transform the analog call to digital and to create VoIP packets. They can have additional features as voice compression, echo cancellation, packets prioritation. Each call is a single IP session transported by a Real-time Transport Protocol (RTP) that runs over UDP.

IP NETWORK

The IP backbone provides the connectivity among the end elements. The IP Network provides smooth delivery of the voice. The IP network must treat voice and data traffic in a different way. To assure good quality of transmission it must be able to prioritize the different traffic types. Classic Circuitswitching telecommunications dedicate channels, reserving bandwidth as it is needed out of the trunk links interconnecting the switches. For example, a phone conversation reserves a single DS-0 channel, and that end-to-end connection is used only for the single conversation. IP networks are quite different from the circuit-switch infrastructure in that it is a packet-network, and it is based on the idea of needed availability. Class of service ensures that packets of a specific application are given priority. This prioritization is required for real-time VoIP applications to ensure that the voice service is unaffected by other traffic flows.

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What equipment is needed for VoIP?

1)Broadband connection

2)voip phone

3)nexton softswitches

4)router

5)audiocodec

6)astric server

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What is voip?

Voip stands for voice over internet protocol. It means the transmission of voice and call control data over the internet.
It also referred to as ip telephony or internet telephony.

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Answers

How does SIP support caller ID?

Ans: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.

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Answers

1.Does SIP support the standard telephone features?

Ans:Yes. SIP supports, among others: 1.call forwarding unconditional, busy, ...
2.call transfer (call control spec)

3.caller ID

4.call hold

5.3-way conferences and multiparty conferencing (call control spec)

6.call return ("*69")

7.call park (with NOTIFY)

8.follow-me

9.find-me

10.call waiting

11.IVR systems

12.multiple line presences

13.call waiting

14.camp on

15.call queueing

16.automatic call distribution

17.do not disturb

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Wednesday, July 13, 2011

Relationship to Other Protocols

#Does SIP do conference control?

#What is the relationship between
MGCP/Megaco/H.GCP and SIP?

#What is SIP+ and how does it
relate to SIP?

#How does SIP compare to H.323?

#Can SIP be used for Internet
telephony gateways (ITGs)?

#Can H.323 and SIP be used
together?

#How do I interconnect Q.931
(ISDN signaling) and SIP?

#How do I interconnect ISUP (SS7
signaling) and SIP?

#What are the different addresses
in SIP?

#How do I put a call on hold?
What is sip-cgi and how does it
relate to CPL?

#Is there a SIP interoperability
certification? How can I test
interoperability with others?

SIP ProtocolOperation

1.What does the [H14.17] in RFC
2543 stand for?


2.Do callers need to know the
location of the location server?

3.Which parts of SIP are case-
sensitive or case-insensitive?

4.What is the difference between a
call leg and a call id?

5.What is the difference between
tag and branch-id?

6.How can one recognize a
retransmitted request?

7.How does a caller find its local
registrar?

8.Is the domain of the request-URI
and the To header always the
same?

9.Are ACK requests retransmitted?

10.How are BYE requests routed?

11.Can I CANCEL requests other
than the first INVITE?

12.What is the relationship between
the From, Contact, Via and
Record-Route/Route headers?

13.How are URLs compared?

14.What's the difference between
the request URIs tel:
+12125551212 and
sip:12125551212@gw.com?

15.Does SIP do admission control?

16.Does SIP administer bandwidth?

17.Do I always need a proxy or
redirect server?

18.How does a caller find its proxy
server?

19.What's the difference between a
stateless and a stateful proxy
server?

20.Why can a forking SIP proxy not
be stateless?

21.How do I handle the case where
multiple requests are received at
a SIP server, each from a
different upstream proxy?

22.How does a caller find the
remote SIP client of the callee?

23.How does SIP get through a
firewall?

24.What are the issues if SIP is used
behind a NAT?

25.How does SIP do "call progress
tones" or "ring back"?

26.Does SIP do keep-alive?

27.Why does SIP not have a
Content-Transfer-Encoding
header?

28.I want SIP to be more compact.
What can I do?

SIP Functionality

1.Does SIP support the standard
telephone features?

2.How does SIP support caller ID?

3.Should SIP be used to join a
conference from a web page?

4.Can a SIP-initiated session have
zero or one participants?

5.How do I charge/bill for Internet
telephony using SIP?

6.How do prepaid calling cards
work in SIP?

7.Does SIP carry DTMF?

SIP Interview Questions

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?

11.What is Dialog?

12.What is Transaction? How many transaction you can find for INVITE to 200 OK What is Refer?

13.What is the difference between call leg and call id?

14.What is the major difference between stateless and statefull?

15.How to identify call loop?

16.What are the parameters you will check to know call got looped?

17.What is the difference between Route and Record-Route?

18.What is via, Branch, Tag? What is Early dialog?