Basic Questions

1. What is SIP, and why is it used?

o Answer: SIP (Session Initiation Protocol) is a signaling protocol

used to establish, manage, and terminate multimedia
communication sessions such as voice and video calls over IP
networks. It is widely used in VoIP systems for call setup,
modification, and teardown.

2. Explain the basic working of SIP.

o Answer: SIP works by exchanging messages between endpoints

to establish communication. It uses methods like INVITE to
initiate a session, ACK to confirm it, and BYE to terminate it. It
operates at the application layer of the OSI model.

Intermediate Questions

3. What are SIP Methods? Name a few.

o Answer: SIP methods are request types used to perform

different actions. Examples include:

 INVITE: Initiates a session.

 ACK: Confirms a session.

 BYE: Terminates a session.

 REGISTER: Registers a user agent with a SIP server.

 CANCEL: Cancels a pending request.

 OPTIONS: Queries a server's capabilities.

4. What is the role of a SIP proxy server?

o Answer: A SIP proxy server acts as an intermediary to route SIP

messages between clients. It helps in call establishment,
 registration, authentication, and forwarding SIP requests and
responses.

5. What are the primary components of a SIP system?

o Answer:

 User Agent (UA): Endpoints that initiate or receive calls.

 Registrar: Manages user registration and location

services.

 Proxy Server: Forwards SIP messages and handles

signaling.

 Redirect Server: Directs clients to an alternate server or

endpoint.

 Session Border Controller (SBC): Secures and manages

VoIP traffic.

 What is the difference between a SIP URI and a TEL URI?

 Answer:

o SIP URI: Identifies a SIP endpoint (e.g.,

sip:[email protected]).

o TEL URI: Identifies a telephone number within SIP (tel:

+1234567890), often used in PSTN interworking.

 How does SIP support multimedia sessions?

 Answer: SIP itself is used for signaling and establishing sessions, while
media data is typically transported using other protocols like RTP (Real-
Time Transport Protocol). SIP negotiates session parameters using SDP.

 What is the role of the Contact header in SIP?

 Answer: The Contact header specifies the exact SIP URI where a user
can be reached for the current session. It is used for routing future
requests directly to the endpoint.

 Explain the CSeq header in SIP. Why is it important?

 Answer: The CSeq (Command Sequence) header identifies and orders

SIP transactions, ensuring correct request and response correlation.
 What is SIP forking, and how is it handled?

 Answer: SIP forking occurs when a single INVITE request is sent to

multiple endpoints. This is managed by the SIP proxy server, which
forwards the request and aggregates responses. The first 200 OK
response usually establishes the session.

Advanced Questions

6. Explain the difference between SIP and H.323.

o Answer:

 SIP: Text-based protocol, simpler, and more extensible.

Operates at the application layer.

 H.323: Binary-based protocol, more complex, includes

multiple protocols within its suite for multimedia
communication. Operates at the network layer.

7. How does SIP handle NAT traversal issues?

o Answer: SIP handles NAT traversal using techniques like:

 STUN (Session Traversal Utilities for NAT): Determines

the public IP and port of a device.

 TURN (Traversal Using Relays around NAT): Relays

media traffic through a server.

 ICE (Interactive Connectivity Establishment):

Combines STUN and TURN to find the best path.

8. What are SIP responses, and how are they categorized?

o Answer: SIP responses are status codes sent in reply to SIP

requests. Categories include:

 1xx (Provisional): Informational (e.g., 180 Ringing).

 2xx (Success): Request was successful (e.g., 200 OK).

 3xx (Redirection): Further action required (e.g., 302

Moved Temporarily).

 4xx (Client Error): Request failed due to client-side

issues (e.g., 404 Not Found).
  5xx (Server Error): Server-side issues (e.g., 500 Internal
Server Error).

 6xx (Global Error): Cannot fulfill the request globally

(e.g., 603 Decline).

 What is the difference between early media and in-call media in

SIP?

 Answer:

o Early Media: Media exchanged before the call is fully

established, such as ring-back tones.

o In-Call Media: Media exchanged after the session is fully

established (200 OK and ACK have been exchanged).

 How is authentication handled in SIP?

 Answer: SIP uses HTTP Digest Authentication. The server sends a 401
Unauthorized response with a challenge. The client computes a hash of
the username, password, and other details using the challenge and
resends the request.

 What is the difference between SIP over UDP, TCP, and TLS?

 Answer:

o UDP: Lightweight but may lead to packet loss.

o TCP: Reliable, ensures packet delivery.

o TLS: Secure, encrypts SIP messages to prevent interception.

 Explain the difference between stateful and stateless SIP proxy

servers.

 Answer:

o Stateful Proxy: Maintains transaction state and processes

responses.

o Stateless Proxy: Forwards messages without retaining any

transaction state.

 How does SIP work with ENUM?

  Answer: ENUM (Telephone Number Mapping) translates a phone
number into a SIP URI, enabling VoIP communication using existing
telephone numbers.

Scenario-Based Questions

9. How would you troubleshoot a SIP call setup failure?

o Answer: Steps to troubleshoot include:

1. Verify network connectivity between endpoints.

2. Check the INVITE request and response flow using tools like
Wireshark.

3. Ensure correct SIP server registration and DNS resolution.

4. Examine SIP headers (e.g., To, From, Contact) for

misconfiguration.

5. Look for errors in the response codes (e.g., 403 Forbidden,

404 Not Found).

10. What is the purpose of SDP in SIP, and what does it

contain?

o Answer: SDP (Session Description Protocol) is used within SIP

messages to describe multimedia session details. It contains:

 Media types (audio, video).

 Codec information.

 IP address and port number.

 Session timing.

11. A user reports one-way audio during a SIP call. How would you troubleshoot it?
o Answer:
  Verify that RTP packets are being sent and received in both directions
using a network tool (e.g., Wireshark).
 Check NAT and firewall settings for RTP ports.
 Confirm the correct IP address and port in the SDP c= and m= lines.
 Test media relay configurations like STUN, TURN, or ICE.
12. How would you handle a 408 Request Timeout error?
o Answer:
 Verify the SIP client can reach the SIP server.
 Check for network delays or timeouts.
 Analyze DNS resolution for the SIP server.
 Ensure proper registration and session establishment.
13. A SIP call fails, and you receive a 403 Forbidden response. What could be the
reasons?
o Answer:
 Authentication failure due to incorrect credentials.
 The caller is not authorized to use the requested service.
 IP blocking or blacklist rules on the SIP server.
14. A call setup is taking too long. How do you diagnose this issue?
o Answer:
 Inspect SIP request-response times using a packet capture tool.
 Check for DNS delays in resolving SIP URIs.
 Verify that all intermediate proxy servers are functioning.
 Look for network congestion or packet loss.
15. How would you investigate SIP message retransmissions?
o Answer:
 Examine the network for packet loss or latency.
 Ensure proper handling of SIP Via and Call-ID headers.
 Check the transport layer (UDP/TCP) for reliability issues.
16. If a SIP session cannot be established due to codec mismatch, what steps would you
take?
o Answer:
 Analyze the SDP offer and answer to identify mismatched codecs.
 Verify that both endpoints support at least one common codec.
 Check SIP server configurations to ensure codec negotiation is enabled.
17. What would you do if a SIP device cannot register with the server?
o Answer:
 Check network connectivity to the SIP server.
 Verify SIP account credentials (username, password).
 Inspect the REGISTER request and server response using a packet analyzer.
 Confirm that the SIP server is operational and reachable.