20 SIP Protocol Interview Questions and Answers

The Session Initiation Protocol is a signaling protocol used for initiating, maintaining, and terminating real-time sessions that include voice, video, and messaging applications. SIP is widely used in VoIP telephony systems and can be used to establish two-way or multi-party communications. If you’re interviewing for a position that involves VoIP or telephony systems, you may be asked questions about SIP. In this article, we’ll review some common SIP questions and how you should answer them.

SIP Protocol Interview Questions and Answers

Here are 20 commonly asked SIP Protocol interview questions and answers to prepare you for your interview:

1. What is the SIP protocol?

The SIP protocol is a communication protocol that is commonly used for VoIP (Voice over IP) applications. It is responsible for setting up, maintaining, and tearing down communication sessions between two or more devices.

2. What are some common use cases for SIP?

SIP is most commonly used for VoIP (Voice over IP) applications, but can also be used for other types of communication such as video conferencing or instant messaging.

3. Why do we need a SIP proxy server?

A SIP proxy server is used to help route SIP requests to the appropriate destination. It can also be used to provide features such as call forwarding and call waiting. In addition, a SIP proxy server can provide security features such as authentication and encryption.

4. Is it possible to enable multiple registrations from different IP addresses on a single SIP account? If yes, then how?

Yes, it is possible to enable multiple registrations from different IP addresses on a single SIP account. To do this, you will need to create a separate SIP URI for each IP address that you want to register. Each SIP URI will need to be registered with the SIP server individually.

5. Can you explain what SIP trunking is and why it’s used?

SIP trunking is a VoIP technology that uses the SIP protocol to connect an on-premises PBX system to the public switched telephone network (PSTN). SIP trunking can provide significant cost savings over traditional PSTN lines, as well as offer additional features and flexibility.

6. Is it possible to upgrade an existing VoIP phone system to support session border controllers? If yes, then how?

Yes, it is possible to upgrade an existing VoIP phone system to support session border controllers. This can be done by installing a SIP proxy server on the network and configuring the VoIP phone system to use it. The SIP proxy server will then act as a session border controller, allowing the VoIP phone system to communicate with other VoIP systems across the network.

7. Can you explain in detail how two systems can establish a call using SIP?

In order to establish a call using SIP, the two systems need to first exchange some basic information in order to set up the call. This includes exchanging information about the type of call that is being made, the location of the two systems, and the IP addresses that will be used to communicate. Once this information has been exchanged, the two systems can start the process of establishing a call. This involves sending signals back and forth in order to establish a connection between the two systems. Once a connection has been established, the two systems can start communicating with each other.

8. How does message routing work with SIP?

SIP uses a process called message routing to determine where a message should be sent next. Message routing is based on the message headers, which contain information about the sender, recipient, and the route the message has taken so far. SIP uses this information to determine where the message should be sent next, and it can also use it to determine if the message should be forwarded to another server or delivered directly to the recipient.

9. Can you explain what a Session Border Controller (SBC) is in context of SIP?

A Session Border Controller is a device that helps to manage SIP sessions. It can be used to provide security, to help with NAT traversal, and to provide other functions.

10. Can you give me a detailed explanation of NAT traversal and its working in relation to SIP?

NAT traversal is a technique used to allow devices behind a NAT firewall to communicate with each other. This is often necessary when using VoIP applications, since VoIP uses UDP which is often blocked by NAT firewalls. NAT traversal allows the VoIP application to use a different port which is not blocked by the NAT firewall.

11. How do you configure presence subscriptions in Asterisk so that they adhere to RFC 3856 recommendations?

You can configure presence subscriptions in Asterisk by adding the following line to the sip.conf file:

subscribecontext=presence

This will ensure that presence subscriptions adhere to RFC 3856 recommendations.

12. What is the difference between SDP, H264-SVC, VP8, and WebRTC?

SDP is a session description protocol that is used to describe multimedia sessions for the purposes of session establishment, announcement, and modification. H264-SVC is a video codec that supports scalable video coding, meaning that it can be used to encode video at different resolutions and bitrates. VP8 is a video codec that is designed for use with the WebRTC protocol. WebRTC is a real-time communication protocol that supports audio and video streaming.

13. What is ICE and TURN? Can you explain their usage?

ICE is a protocol that helps two devices connect to each other over the internet, even if they are behind NATs. TURN is a protocol that helps when ICE fails, by relaying traffic between the two devices.

14. Are there any security risks associated with SIP? If yes, then how can they be overcome?

Yes, there are some security risks associated with SIP, but they can be overcome by using a secure version of the protocol, such as SIPS. SIPS uses Transport Layer Security (TLS) to encrypt SIP messages, which prevents eavesdropping and man-in-the-middle attacks.

15. What is your opinion on the quality of SIP compared to other protocols such as Skype or Whatsapp?

I believe that SIP is a high quality protocol that offers a number of advantages over other protocols. For example, SIP is designed to be very scalable, so it can easily be used by large organizations with many users. Additionally, SIP is very flexible and can be easily customized to meet the needs of specific organizations.

16. Should I consider using TLS for my SIP infrastructure? Why/why not?

TLS can provide a number of benefits for a SIP infrastructure, including improved security and increased reliability. However, it can also add complexity and overhead, so it is important to weigh the pros and cons carefully before deciding whether or not to use TLS.

17. How does SIP differ from HTTP?

SIP is a communication protocol that is used for signaling and controlling multimedia communication sessions. HTTP, on the other hand, is a application protocol that is used for transferring data between clients and servers. SIP is designed to be independent of the underlying transport layer, while HTTP relies on TCP. SIP also uses a different message format than HTTP.

18. Can you explain how signaling works in SIP?

SIP is a signaling protocol that is used to set up and tear down communication sessions. In order to do this, SIP uses a variety of methods to signal the other party or parties involved in the session. These methods can include INVITE, CANCEL, BYE, and OPTIONS.

19. What is the best way to implement UDP load balancing for SIP servers?

There are a few different ways to implement UDP load balancing for SIP servers, but the best way will likely depend on your specific needs and setup. One popular method is to use a software load balancer, which can be configured to distribute traffic evenly across a group of servers. Another option is to use a hardware load balancer, which can provide more granular control over traffic distribution.

20. How does authentication happen in SIP?

Authentication in SIP can happen in a few different ways, but the most common is through the use of a username and password. When a SIP client tries to connect to a SIP server, the server will challenge the client to provide a valid username and password. If the client can provide the correct credentials, then the server will allow the connection to be made.