SIP Session Initiation Protocol Certification Exam — Questions and Answers

Question 1: What is the purpose of the SIP SUBSCRIBE method?

Establishes a permanent monitoring session
Registers a UA with a registrar for a fixed duration
Subscribes to a media multicast group
Requests notification of events from a notifier (Correct answer)

Correct answer: Requests notification of events from a notifier

SUBSCRIBE is used in the SIP event notification framework (RFC 6665) to request that a notifier send NOTIFY messages when a specific event occurs.

Question 2: In a SIP 'call hijacking via re-INVITE' attack, what is the attacker's goal?

Prevent codec renegotiation to degrade call quality
Terminate the call by sending a spoofed BYE
Steal authentication credentials from the re-INVITE transaction
Redirect the media stream to an attacker-controlled endpoint by modifying SDP in a re-INVITE (Correct answer)

Correct answer: Redirect the media stream to an attacker-controlled endpoint by modifying SDP in a re-INVITE

By injecting a forged re-INVITE with modified SDP, an attacker can redirect the RTP media stream to their own endpoint, effectively intercepting the audio.

Question 3: What is the importance of communication skills in Security and NAT Traversal?

Communication skills cannot be learned
Clear communication prevents errors, builds relationships, and ensures understanding among all parties (Correct answer)
Communication is only important for managers
Technical skills are sufficient without communication

Correct answer: Clear communication prevents errors, builds relationships, and ensures understanding among all parties

Effective communication is essential for coordination, error prevention, and stakeholder relationships.

Question 4: What is the effect of an INVITE glare condition in SIP session management?

A proxy resolves the conflict by choosing one re-INVITE
Both UAs send a re-INVITE simultaneously, causing each to receive a 491 and retry after a random delay (Correct answer)
The session terminates automatically when glare is detected
Both re-INVITEs are queued and processed sequentially

Correct answer: Both UAs send a re-INVITE simultaneously, causing each to receive a 491 and retry after a random delay

Glare occurs when both endpoints send re-INVITEs at the same time; each responds 491, and both must back off and retry after randomized timers.

Question 5: Which SIP response code does a UAS return when a session timer refresh interval is too short?

422 Session Interval Too Small (Correct answer)
480 Temporarily Unavailable
416 Unsupported URI Scheme
491 Request Pending

Correct answer: 422 Session Interval Too Small

422 Session Interval Too Small (RFC 4028) is returned with a Min-SE header indicating the minimum acceptable refresh interval.

Question 6: How does a UAC properly terminate a confirmed SIP dialog (active call)?

By sending a BYE request within the existing dialog (Correct answer)
By closing the underlying TCP or UDP connection
By sending a CANCEL request to the UAS
By sending a REGISTER with an Expires header value of 0

Correct answer: By sending a BYE request within the existing dialog

A confirmed dialog is terminated by sending a BYE request within the dialog; CANCEL can only be used to terminate a pending INVITE transaction before confirmation.

Question 7: What role does continuing education play in SIP Architecture?

Keeping practitioners current with evolving best practices and new developments (Correct answer)
It is optional and rarely helpful
Only beneficial for management
Only required for new practitioners

Correct answer: Keeping practitioners current with evolving best practices and new developments

Continuing education ensures practitioners stay current with advances in their field.

Question 8: What makes a professional competent in Call Setup and Teardown?

Only certifications matter
Only experience matters
A combination of education, practical experience, ongoing learning, and adherence to professional standards (Correct answer)
Only formal education matters

Correct answer: A combination of education, practical experience, ongoing learning, and adherence to professional standards

Competence requires the integration of knowledge, skills, experience, and professional conduct.

Question 9: How does technology impact modern practices in SIP Architecture?

Technology improves efficiency, accuracy, and access to information while requiring new skills (Correct answer)
Technology has no impact on this field
Technology replaces all human judgment
Technology is only useful for record keeping

Correct answer: Technology improves efficiency, accuracy, and access to information while requiring new skills

Technology enhances capabilities while also requiring practitioners to develop new competencies.

Question 10: Which SIP security mechanism uses the 'Identity' header to provide cryptographic proof of caller identity?

TLS mutual authentication
S/MIME signing
RFC 8224 (STIR) (Correct answer)
Digest Authentication

Correct answer: RFC 8224 (STIR)

RFC 8224 (STIR - Secure Telephone Identity Revisited) uses the Identity header containing a JWT token to cryptographically assert caller identity.

Question 11: What does a SIP response code of 180 Ringing indicate?

The destination endpoint is currently ringing (Correct answer)
The call has been successfully established
The request cannot be fulfilled due to an error
The request has been received and is being processed

Correct answer: The destination endpoint is currently ringing

A SIP response code of 180 Ringing is a provisional response indicating that the called party's user agent has received the INVITE and is currently alerting the user. It informs the calling party that the call is progressing and the destination is being notified, often accompanied by a ringback tone. This response does not establish the call but signals that the recipient is being alerted.

Question 12: What does the SIP 'To' header field represent in an initial request?

The logical recipient or callee of the request (Correct answer)
The transport protocol to use
The IP address of the proxy server
The session expiration time

Correct answer: The logical recipient or callee of the request

The 'To' header identifies the logical recipient of the request, typically the callee's address-of-record (AOR).

Question 13: When a SIP proxy performs Record-Routing, what is the primary reason?

To encrypt all subsequent requests in the dialog
To cache media packets for quality monitoring
To remain in the signaling path for all subsequent requests within the dialog (Correct answer)
To prevent REGISTER requests from leaving the domain

Correct answer: To remain in the signaling path for all subsequent requests within the dialog

Record-Routing ensures the proxy stays in the signaling path for all in-dialog requests such as re-INVITEs and BYEs.

Question 14: Which component of a SIP transaction is responsible for maintaining state between a client request and the server response?

SIP Dialog (Correct answer)
SIP Registrar
SIP Proxy
SIP Gateway

Correct answer: SIP Dialog

A SIP Dialog is a peer-to-peer relationship between two SIP user agents that persists for the duration of a call or session. It maintains state information, such as sequence numbers and route sets, between a client request and subsequent server responses. This stateful connection ensures that related messages are correctly associated and processed within the context of an ongoing session, providing continuity.

Question 15: What is the primary purpose of a SIP transaction?

To establish a persistent connection between two endpoints
To represent a single request and all responses to that request (Correct answer)
To maintain session state for the duration of a call
To route SIP messages across multiple proxies

Correct answer: To represent a single request and all responses to that request

A SIP transaction consists of a single request and any responses to that request, terminated by a final response, as defined in RFC 3261.

Question 16: What role does continuing education play in Call Setup and Teardown?

Keeping practitioners current with evolving best practices and new developments (Correct answer)
Only required for new practitioners
Only beneficial for management
It is optional and rarely helpful

Correct answer: Keeping practitioners current with evolving best practices and new developments

Continuing education ensures practitioners stay current with advances in their field.

Question 17: What is the role of a 'Back-to-Back User Agent' (B2BUA) from a security perspective?

It relays SIP messages transparently without inspecting content
It provides end-to-end encryption between two UACs
It terminates and re-originates SIP calls, providing a security policy enforcement point (Correct answer)
It authenticates users using certificate-based identity

Correct answer: It terminates and re-originates SIP calls, providing a security policy enforcement point

A B2BUA terminates and re-originates SIP sessions, allowing it to enforce security policies, strip sensitive headers, normalize content, and prevent malformed messages from propagating.

Question 18: What is the most important foundational concept in Call Setup and Teardown?

Memorizing all rules without understanding
Ignoring established guidelines
Understanding core principles and their practical application in Call Setup and Teardown (Correct answer)
Only following instructions from supervisors

Correct answer: Understanding core principles and their practical application in Call Setup and Teardown

Core principles provide the foundation for all decision-making and practice in Call Setup and Teardown.

Question 19: What is the primary purpose of standardized procedures in SIP Architecture?

To slow down work processes
To benefit only management
To create unnecessary bureaucracy
To ensure consistency, safety, and quality across all practitioners (Correct answer)

Correct answer: To ensure consistency, safety, and quality across all practitioners

Standardized procedures ensure that all practitioners deliver consistent, safe, and high-quality outcomes.

Question 20: How does SIP interoperability with H.323 typically get managed?

By using a SIP proxy server
By avoiding the use of H.323 in SIP environments
Through a protocol gateway that translates between SIP and H.323 (Correct answer)
By direct communication between SIP and H.323 endpoints

Correct answer: Through a protocol gateway that translates between SIP and H.323

SIP and H.323 are two distinct and complex signaling protocols for multimedia communication, each with its own architecture, message formats, and call flow procedures. Direct communication between endpoints using these different protocols is not possible without translation. Therefore, a protocol gateway is required to translate the signaling messages and media parameters between SIP and H.323, enabling interoperability between the two environments.

Question 21: A SIP B2BUA differs from a SIP proxy primarily because:

A B2BUA only modifies the Via header
A B2BUA operates only on UDP transport
A B2BUA terminates and re-originates each SIP dialog, maintaining full control of both legs (Correct answer)
A B2BUA cannot handle INVITE transactions

Correct answer: A B2BUA terminates and re-originates each SIP dialog, maintaining full control of both legs

A B2BUA (Back-to-Back User Agent) acts as a UAS on one side and a UAC on the other, terminating and re-originating dialogs, giving it full control over both call legs.

Question 22: To ensure secure communication in SIP, which of the following practices is recommended?

Use SIP over plain UDP for faster communication
Implement SIP over TLS and use secure authentication methods (Correct answer)
Avoid using authentication to reduce complexity
Disable encryption to simplify configuration

Correct answer: Implement SIP over TLS and use secure authentication methods

To ensure secure communication in SIP, the recommended practice is to implement SIP over TLS (Transport Layer Security) and utilize robust authentication methods like Digest Authentication. TLS encrypts the SIP signaling messages, protecting them from eavesdropping and tampering, while secure authentication verifies the identity of communicating parties. This combined approach significantly enhances the confidentiality, integrity, and authenticity of SIP communications.

Question 23: When a SIP trunk interconnects two enterprises with overlapping private numbering plans, which element typically resolves number translation?

DNS SRV record
STUN server
Session Border Controller with dial plan translation (Correct answer)
SIP Registrar

Correct answer: Session Border Controller with dial plan translation

A Session Border Controller (SBC) with dial plan translation rewrites numbers to avoid conflicts between overlapping private numbering plans.

Question 24: What are the key regulatory requirements affecting Call Setup and Teardown?

There are no regulations
Regulations are optional guidelines
Only federal regulations apply
Compliance with applicable laws, industry standards, and licensing requirements specific to the field (Correct answer)

Correct answer: Compliance with applicable laws, industry standards, and licensing requirements specific to the field

Practitioners must comply with all applicable regulations, which may include federal, state, and industry-specific requirements.

Question 25: Which combination of headers uniquely identifies a SIP dialog?

From URI and To URI only
Call-ID and CSeq number
Via branch parameter and Call-ID
Call-ID, From tag, and To tag (Correct answer)

Correct answer: Call-ID, From tag, and To tag

RFC 3261 defines a SIP dialog as uniquely identified by the three-tuple of Call-ID, From tag, and To tag.

Question 26: Which SIP extension supports the exchange of instant messages between users?

SIP-INVITE
SIP-IM (Correct answer)
SIP-OPTIONS
SIP-CALL

Correct answer: SIP-IM

SIP is a versatile protocol that can be extended beyond traditional voice and video calls to support other real-time communication features. The SIP-IM (Instant Messaging) extension specifically defines how text-based instant messages are encapsulated and exchanged using SIP signaling. This allows for integrated messaging capabilities within the SIP framework.

Question 27: How should ethical dilemmas be handled in Call Setup and Teardown?

Make decisions based solely on personal preference
Ignore ethical concerns
Follow established ethical guidelines, consult with supervisors, and prioritize stakeholder welfare (Correct answer)
Always choose the cheapest option

Correct answer: Follow established ethical guidelines, consult with supervisors, and prioritize stakeholder welfare

Ethical decision-making requires following professional guidelines and prioritizing the welfare of all stakeholders.

Question 28: How does technology impact modern practices in Call Setup and Teardown?

Technology is only useful for record keeping
Technology has no impact on this field
Technology improves efficiency, accuracy, and access to information while requiring new skills (Correct answer)
Technology replaces all human judgment

Correct answer: Technology improves efficiency, accuracy, and access to information while requiring new skills

Technology enhances capabilities while also requiring practitioners to develop new competencies.

Question 29: In SIP, what does 'SPIT' stand for and what threat does it represent?

SIP Protocol Interception Technique — call eavesdropping
Spam over Internet Telephony — unsolicited bulk voice calls (Correct answer)
Session Protocol Integrity Testing — security auditing
Secure Peer Identity Token — caller verification

Correct answer: Spam over Internet Telephony — unsolicited bulk voice calls

SPIT (Spam over Internet Telephony) refers to unsolicited bulk calls over VoIP, analogous to email spam but for voice communications.

Question 30: What does a SIP 302 Moved Temporarily response instruct the UAC to do?

Authenticate with the proxy
Terminate the session immediately
Resend the request to an alternate URI provided in the Contact header (Correct answer)
Retry the request after a delay

Correct answer: Resend the request to an alternate URI provided in the Contact header

302 Moved Temporarily provides an alternate Contact URI and instructs the UAC to retry the request at that new address.

Question 31: How should risk be managed in SIP Architecture?

By avoiding all risk entirely
Risk management is not necessary
By ignoring potential problems
Through identification, assessment, mitigation strategies, and ongoing monitoring of potential risks (Correct answer)

Correct answer: Through identification, assessment, mitigation strategies, and ongoing monitoring of potential risks

Effective risk management involves a systematic process of identifying, evaluating, and addressing potential threats.

Question 32: Which SIP authentication mechanism involves sending a challenge-response pair to verify user credentials?

Digest Authentication (Correct answer)
OAuth Authentication
Basic Authentication
Kerberos Authentication

Correct answer: Digest Authentication

Digest Authentication is a common SIP authentication mechanism that involves a challenge-response process to verify user credentials without sending the password in plain text. The server sends a challenge, and the client responds with a hash of the password, nonce, and other information. This method provides a more secure alternative to Basic Authentication by protecting credentials from eavesdropping.

Question 33: Which SIP mechanism allows a UA to detect if a session is still active when no media is flowing?

Registration refresh
180 Ringing keepalive
Session timers (RFC 4028) (Correct answer)
PRACK retransmission

Correct answer: Session timers (RFC 4028)

RFC 4028 session timers use periodic re-INVITEs or UPDATE requests to verify both sides are still alive.

Question 34: When integrating SIP with Microsoft Teams Direct Routing, the Session Border Controller must support which audio codec mandatorily?

Opus
G.729
G.722
G.711 (PCMU/PCMA) (Correct answer)

Correct answer: G.711 (PCMU/PCMA)

Microsoft Teams Direct Routing requires the SBC to support G.711 (PCMU and PCMA) as a mandatory baseline codec for interoperability.

Question 35: What SIP method is typically used to modify an existing session, such as changing media parameters?

BYE
OPTIONS
ACK
RE-INVITE (Correct answer)

Correct answer: RE-INVITE

The RE-INVITE method is a special form of the INVITE method used to modify an existing, established SIP session without tearing it down. It allows endpoints to change session parameters, such as adding or removing media streams, changing codecs, or updating IP addresses. This provides flexibility for dynamic session management, enabling adjustments to an ongoing call's characteristics.

Question 36: What quality metrics are most important in Call Setup and Teardown?

Measurable outcomes, process compliance, and stakeholder satisfaction (Correct answer)
Only financial metrics
Only speed of completion
No metrics are needed

Correct answer: Measurable outcomes, process compliance, and stakeholder satisfaction

Quality in Call Setup and Teardown is measured through multiple dimensions including outcomes, processes, and satisfaction.

Question 37: Which approach is most effective for problem-solving in Call Setup and Teardown?

Always asking someone else
Ignoring the problem until it resolves itself
Random trial and error
Systematic analysis followed by evidence-based decision making (Correct answer)

Correct answer: Systematic analysis followed by evidence-based decision making

Systematic, evidence-based approaches lead to more reliable solutions than ad hoc methods.

Question 38: What is the importance of communication skills in Call Setup and Teardown?

Clear communication prevents errors, builds relationships, and ensures understanding among all parties (Correct answer)
Communication is only important for managers
Communication skills cannot be learned
Technical skills are sufficient without communication

Correct answer: Clear communication prevents errors, builds relationships, and ensures understanding among all parties

Effective communication is essential for coordination, error prevention, and stakeholder relationships.

Question 39: A SIP UA sends an INVITE to a PSTN gateway that supports only G.711. The UA's SDP offer includes G.729 and G.711. What determines which codec is used?

The UA always wins the codec negotiation
RTCP negotiates the codec after the call is established
SDP answer from the gateway selects the mutually supported codec (Correct answer)
The gateway selects the first codec in the offer

Correct answer: SDP answer from the gateway selects the mutually supported codec

The gateway's SDP answer selects one or more codecs from the offer that it supports, finalizing the codec choice.

Question 40: In SIP, what is an 'address-of-record' (AOR)?

A SIP URI that serves as a public address for a user (Correct answer)
The FQDN of the SIP proxy
The MAC address bound to a SIP phone
The IP address of the UA's NIC

Correct answer: A SIP URI that serves as a public address for a user

An AOR is a SIP URI (e.g., sip:alice@example.com) that represents a user's persistent public identity, independent of their device location.

Question 41: What is the SIP loose routing mechanism and which header does it use?

Source routing via the Via header
Strict routing via the Request-URI
Loose routing via the Route header with 'lr' parameter (Correct answer)
Policy routing via the P-Asserted-Identity header

Correct answer: Loose routing via the Route header with 'lr' parameter

Loose routing (RFC 3261) uses the Route header with the 'lr' parameter, allowing proxies to forward requests without replacing the Request-URI.

Question 42: What are the key regulatory requirements affecting SIP Architecture?

There are no regulations
Compliance with applicable laws, industry standards, and licensing requirements specific to the field (Correct answer)
Only federal regulations apply
Regulations are optional guidelines

Correct answer: Compliance with applicable laws, industry standards, and licensing requirements specific to the field

Practitioners must comply with all applicable regulations, which may include federal, state, and industry-specific requirements.

Question 43: What is Timer B in SIP INVITE client transactions?

The retransmission interval for INVITE requests over UDP
The time a server waits before sending a 100 Trying response
The maximum time a UAC waits for a final response to an INVITE before timing out (Correct answer)
The interval between sending ACK and cleaning up the transaction

Correct answer: The maximum time a UAC waits for a final response to an INVITE before timing out

Timer B is the INVITE client transaction timeout set to 64*T1 (32 seconds by default), after which the transaction is abandoned if no final response is received.

Question 44: Which header uniquely identifies a SIP transaction?

From tag
Via branch parameter (Correct answer)
Call-ID
CSeq number

Correct answer: Via branch parameter

The Via branch parameter (starting with 'z9hG4bK') uniquely identifies a SIP transaction per RFC 3261.

Question 45: Which attack targets SIP infrastructure by sending a high volume of crafted SIP messages to exhaust server resources?

Eavesdropping
SPIT (Spam over Internet Telephony)
Toll fraud
SIP flooding (DoS) (Correct answer)

Correct answer: SIP flooding (DoS)

SIP flooding attacks send massive volumes of SIP messages to overwhelm a server's processing capacity, constituting a Denial of Service attack.

Question 46: In SIP digest authentication, what is the purpose of the 'nonce' value?

Prevents replay attacks by ensuring each challenge is unique (Correct answer)
Encrypts the user password
Identifies the authentication realm
Specifies the hashing algorithm to use

Correct answer: Prevents replay attacks by ensuring each challenge is unique

The nonce is a server-generated unique value included in the WWW-Authenticate challenge to ensure each authentication response is unique and cannot be reused.

Question 47: What makes a professional competent in SIP Architecture?

A combination of education, practical experience, ongoing learning, and adherence to professional standards (Correct answer)
Only certifications matter
Only formal education matters
Only experience matters

Correct answer: A combination of education, practical experience, ongoing learning, and adherence to professional standards

Competence requires the integration of knowledge, skills, experience, and professional conduct.

Question 48: What does the SIP 'Expires' header control when used in a REGISTER request?

The duration for which the contact binding is valid (Correct answer)
The timeout before a re-INVITE must be sent
The lifetime of the SIP dialog
The maximum session duration for voice calls

Correct answer: The duration for which the contact binding is valid

The Expires header in a REGISTER request tells the registrar how long (in seconds) to keep the AOR-to-contact binding before it expires.

Question 49: Which SIP timer governs how long a UAC waits for a final response before retransmitting an INVITE?

Timer D
Timer A
Timer B (Correct answer)
Timer C

Correct answer: Timer B

Timer B is the INVITE transaction timeout timer; if no final response is received within 64*T1, the transaction times out.

Question 50: What is the purpose of the 'opaque' route set maintained by a UAC after dialog establishment?

To record the authentication credentials for the dialog lifetime
To cache the IP address of the remote UAS for direct media delivery
To store the codec list negotiated during SDP offer/answer
To ensure subsequent requests within the dialog are routed through the same set of proxies (Correct answer)

Correct answer: To ensure subsequent requests within the dialog are routed through the same set of proxies

The route set, built from Record-Route headers received during dialog establishment, ensures all subsequent in-dialog requests follow the same proxy path as the initial INVITE.

Question 51: Which SIP method is used to send mid-session information that does not change the session state?

UPDATE
MESSAGE
INFO (Correct answer)
NOTIFY

Correct answer: INFO

INFO carries application-level information (e.g., DTMF tones) within an established session without modifying the session state.

Question 52: Which SIP header limits the number of proxies that can forward a request?

Max-Hops
TTL
Route-Limit
Max-Forwards (Correct answer)

Correct answer: Max-Forwards

The Max-Forwards header starts at a value (typically 70) and is decremented by each proxy; reaching 0 causes a 483 Too Many Hops response.

Question 53: What quality metrics are most important in SIP Architecture?

Only financial metrics
No metrics are needed
Measurable outcomes, process compliance, and stakeholder satisfaction (Correct answer)
Only speed of completion

Correct answer: Measurable outcomes, process compliance, and stakeholder satisfaction

Quality in SIP Architecture is measured through multiple dimensions including outcomes, processes, and satisfaction.

Question 54: Which threat does the 'Record-Route' header manipulation pose in SIP security?

An attacker can modify codec negotiation during the session
An attacker can reroute the initial INVITE to a different destination
An attacker can insert themselves as a persistent on-path proxy for all in-dialog requests (Correct answer)
An attacker can prevent BYE messages from terminating a call

Correct answer: An attacker can insert themselves as a persistent on-path proxy for all in-dialog requests

By injecting a malicious URI into the Record-Route header, an attacker can ensure all subsequent in-dialog requests (like re-INVITEs and BYE) pass through their proxy.

Question 55: What is the function of the CSeq header in SIP session management?

Identifies the transaction method and sequence number within a dialog (Correct answer)
Stores the session expiration time
Encrypts the session media stream
Routes requests through proxy servers

Correct answer: Identifies the transaction method and sequence number within a dialog

CSeq provides a command sequence number paired with the method name, used to order requests and match responses within a dialog.

Question 56: What is the significance of documentation in SIP Architecture?

Only management needs to document
Documentation is unnecessary busy work
Only required during audits
It provides a record for accountability, quality assurance, and legal compliance (Correct answer)

Correct answer: It provides a record for accountability, quality assurance, and legal compliance

Proper documentation supports accountability, enables quality review, and satisfies legal requirements.

Question 57: Which transport protocol is recommended for SIP interoperability when traversing firewalls and NATs due to its reliable delivery?

UDP
SCTP
TCP (Correct answer)
RTP

Correct answer: TCP

TCP is preferred for firewall and NAT traversal because stateful firewalls track TCP connections, and TCP ensures reliable SIP message delivery.

Question 58: Which IETF RFC defines the SIP-T framework for carrying ISUP information within SIP messages?

RFC 4474
RFC 3711
RFC 3261
RFC 3372 (Correct answer)

Correct answer: RFC 3372

RFC 3372 defines SIP-T, which enables ISUP signaling to be encapsulated within SIP for PSTN interworking.

Question 59: In SIP, what does a 491 Request Pending response indicate?

A re-INVITE or UPDATE was rejected because another transaction is already in progress within the dialog (Correct answer)
Authentication credentials are pending verification
The proxy queue is full and the request must be retried
The requested URI cannot be resolved

Correct answer: A re-INVITE or UPDATE was rejected because another transaction is already in progress within the dialog

491 is returned when a UA receives a re-INVITE while it has a pending unacknowledged re-INVITE, preventing overlapping transactions.

Question 60: What is the importance of communication skills in SIP Architecture?

Technical skills are sufficient without communication
Clear communication prevents errors, builds relationships, and ensures understanding among all parties (Correct answer)
Communication is only important for managers
Communication skills cannot be learned

Correct answer: Clear communication prevents errors, builds relationships, and ensures understanding among all parties

Effective communication is essential for coordination, error prevention, and stakeholder relationships.

SIP Session Initiation Protocol Certification Exam

The SIP Session Initiation Protocol Certification Exam validates expertise in SIP protocol fundamentals, session management, security, interoperability, architecture, and call setup and teardown procedures for VoIP and unified communications professionals.

Questions

90 min

Time Limit

70%

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