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300-815 Cisco Practice Test Questions and Exam Dumps

Question No 1:

The administrator of ABC company is troubleshooting a one-way audio issue for a call using the H.323 protocol in slow-start mode. The administrator asks you to provide the IP and port information for the Real-Time Transport Protocol (RTP) traffic involved in the one-way audio call. You gather the H.225 and H.245 messages for the call in question. 

Where can you locate the RTP IP and port information for both sides of the call? (Note: The call flow does not use media resources like Media Termination Points (MTP) or transcoders.)

Options:

A. H.245 Terminal Capability Set
B. H.245 Open Logical Channel
C. H.225 Connect
D. H.245 Open Logical Channel Acknowledgement

Correct Answer: B. H.245 Open Logical Channel

Explanation:

To troubleshoot the one-way audio issue and identify the RTP IP and port information for the call, we need to understand the role of H.225 and H.245 messages within an H.323 call flow.

H.323 Protocol Overview:

H.323 is a signaling protocol used for multimedia communication, such as voice and video calls. It involves multiple messages exchanged between devices (terminals, gateways, etc.) to establish, maintain, and terminate communication sessions. The H.323 protocol operates alongside other protocols, including the Real-Time Transport Protocol (RTP) for media transmission.

• H.225 is responsible for call signaling and call setup (like call establishment and control).

• H.245 is used to negotiate media capabilities and open logical channels for media streams.

In this scenario, we are interested in locating the IP and port information for the RTP traffic, which carries the actual media (in this case, audio). The RTP IP and port information are typically found in the H.245 Open Logical Channel message.

Key Message Types in the Call Flow:

1. H.245 Terminal Capability Set (A):

• The Terminal Capability Set message is part of the H.245 negotiation process and is used to exchange information about the supported media capabilities between endpoints. This message does not contain the RTP IP and port details. Its primary purpose is to inform the endpoints of the supported codecs and media types, such as audio and video formats, but it doesn’t define the actual media channel for RTP.

2. H.245 Open Logical Channel (B):

• The Open Logical Channel message is sent after the capabilities are exchanged and is used to open a logical media channel for RTP traffic. It includes the essential information to establish the media stream between the two endpoints, including the RTP IP address and port. This is where the RTP session details (IP addresses and port numbers) are specified for both sides of the call.

• The Open Logical Channel message specifies the media types (audio/video) and other transport details necessary to establish the media path, including the RTP parameters.

3. H.225 Connect (C):

• The H.225 Connect message is part of the H.225 signaling process, and its primary purpose is to acknowledge the establishment of the call after initial signaling. This message includes information such as the calling party's IP address and the call reference, but it does not contain RTP-related details. It is mainly concerned with call control and signaling rather than media transport.

4. H.245 Open Logical Channel Acknowledgement (D):

• The Open Logical Channel Acknowledgement message is the acknowledgment response to the Open Logical Channel message. While this message confirms the acceptance of the logical channel setup, it does not contain new RTP IP or port information. It is simply a confirmation that the media channel can be used for RTP traffic.

The correct place to find the RTP IP and port information for the one-way audio call is in the H.245 Open Logical Channel message. This message establishes the media stream between the endpoints and contains the RTP transport parameters (IP address and port), which are crucial for diagnosing media issues such as the one-way audio problem.

Question No 2:

Which two extended capabilities must be configured on dial peers to support fast start-to-early media scenarios in H.323 to SIP interworking? (Choose two.)

A. DTMF
B. BFCP
C. VIDEO
D. FAX
E. AUDIO

Detailed Answer:

The two extended capabilities that must be configured on dial peers for fast start-to-early media scenarios in H.323 to SIP interworking are:

1. AUDIO (E)
   
   

2. VIDEO (C)

Explanation:

In modern VoIP (Voice over IP) systems, interworking between different signaling protocols, such as H.323 and SIP (Session Initiation Protocol), is a critical part of ensuring that multimedia communication between disparate systems works seamlessly. One of the common use cases for interworking between H.323 and SIP is the fast start-to-early media scenario, which facilitates quicker media negotiation and allows media to flow before the call is fully established. In this context, certain capabilities need to be configured correctly on the dial peers to ensure that the media session (audio, video, etc.) starts early in the call setup process.

1. AUDIO (E):

For successful fast start-to-early media negotiation, it is essential to ensure that the audio stream is configured and allowed on both H.323 and SIP dial peers. In many cases, audio is the primary media type in voice communications. By enabling audio on both the H.323 and SIP dial peers, the media session can begin early, bypassing the need for waiting until the signaling process is fully completed. This is particularly important in scenarios where the goal is to provide the user with immediate feedback (e.g., ringing or an interactive voice response system) without waiting for the entire call setup to finish.

2. VIDEO (C):

In addition to audio, video is another extended capability that must be configured to support fast start-to-early media scenarios when video calls are involved. Video calls often require additional negotiation and setup time, but by enabling video on the dial peers in a fast-start scenario, video media can begin flowing as soon as the signaling exchange allows. This ensures a smooth and timely experience for users who expect both voice and video communication to start almost simultaneously. Without proper video configuration on the dial peers, the video portion of the call could be delayed until after the full call setup, which could disrupt the user experience.

Why Other Options Are Not Correct:

• DTMF (A): Dual-tone multi-frequency (DTMF) signaling is used to transmit keypress signals (e.g., for IVR or call routing) but is not directly involved in fast-start-to-early media negotiation. DTMF can be handled separately from the core audio/video media streams.

• BFCP (B): The Binary Floor Control Protocol (BFCP) is used in collaboration systems for controlling access to shared media like video conferencing rooms, but it is not a direct requirement for setting up audio or video media streams in H.323 to SIP interworking scenarios.

• FAX (D): Fax transmission is not a media type commonly associated with fast-start-to-early media scenarios. Fax is typically handled by dedicated protocols (like T.38) or fallback methods, but it is not critical for fast-start-to-early media in the context of H.323 to SIP interworking.

For H.323 to SIP interworking in fast start-to-early media scenarios, the key extended capabilities that must be configured on dial peers are AUDIO and VIDEO. These capabilities ensure that the necessary media streams can begin flowing promptly during call setup, enhancing the user experience by minimizing delays in establishing media communication.

Question No 3:

When troubleshooting H.323 call setup, which message indicates that the called party has been notified about the incoming call?

A. ALERTING
B. PROCEEDING
C. CONNECT
D. RINGING

Answer: A. ALERTING

Explanation:

In H.323 call setup, a series of messages are exchanged between the calling and called parties to establish the connection. These messages correspond to different stages of the call progress. Understanding these messages is crucial for diagnosing issues when troubleshooting H.323 calls.

H.323 Call Setup Process: The process of setting up a call in H.323 typically involves the following steps:

1. Call Initiation: The calling party initiates a call by sending a Setup message to the called party. This message contains the necessary information about the call, such as the calling party's address, the media capabilities, and other call parameters.

2. Proceeding (B): Once the Setup message is received, the called party may respond with a Proceeding message. This message indicates that the called party has received the call setup request and is in the process of determining whether to accept the call or not. However, the Proceeding message does not indicate to the calling party that the called party is aware of the call, only that the call processing is underway.

3. Alerting (A): The Alerting message is sent when the called party has been notified about the incoming call. This is the key message that informs the calling party that the called party's device has started alerting the user (e.g., ringing the phone). The Alerting message doesn't necessarily mean the called party has answered the call, but it indicates that the called party is aware of the call and is being alerted.

4. Ringing (D): Although the Ringing message might seem related to this step, it is typically used in specific environments or implementations (like ISDN) to indicate that the called party's phone is actually ringing. The Alerting message, in contrast, is the standardized message in H.323 that indicates the called party is being notified, which aligns more closely with the intention of the question.
   Connect (C): The Connect message is sent once the called party answers the call. This message marks the successful establishment of the call, and media transmission begins. At this point, the call is fully established, and both parties can communicate. The Connect message signifies the conclusion of the call setup phase.

The key message that informs you that the called party is being notified about the call during H.323 call setup is the Alerting message. This step occurs after the called party has received the initial call setup and is being alerted about the call (e.g., by ringing the phone), but before the call is answered. It is essential for troubleshooting because it shows that the call is in progress, and the recipient has been notified but has not yet answered the call.

Question No 4:

End users at a new site report being unable to hear the remote party during calls with users at headquarters. However, calls to and from the Public Switched Telephone Network (PSTN) function as expected. To investigate the issue related to the Session Initiation Protocol (SIP) signaling, 

Which field in the SIP message could provide a useful clue for troubleshooting?

A. Contact: header of the 200 OK response
B. Allow: header of the 200 OK response
C. o= line of SDP content
D. c= line of SDP content

Answer:

The correct answer is D. c= line of SDP content.

Explanation:

In troubleshooting SIP-based communication issues, particularly those related to call audio (like the scenario described where users cannot hear each other), the key focus often lies in the Session Description Protocol (SDP) content. SIP uses SDP to describe the media capabilities for a call, including the codecs, IP addresses, and ports that should be used for the media stream.

In the scenario provided, where end users at a new site are unable to hear remote parties during calls with users at headquarters, it suggests an issue with media negotiation or the actual delivery of media (i.e., audio). Here's why the c= line of the SDP is the most important field to check:

1. The Role of SDP in SIP Communication:
   
   

• The SIP protocol itself is responsible for setting up, maintaining, and terminating communication sessions. However, it is the SDP within the SIP messages that defines the media parameters like which codecs to use, where to send the audio (IP addresses), and which ports to use.
  
  

• SIP signaling exchanges offer the initial negotiation of the call setup, but once the session is established, the media is transmitted directly between endpoints based on the SDP's information.
  
  

2. The "c=" Line in SDP:
   
   

• The "c=" line of the SDP content specifies the connection information, such as the IP address and port number that the calling party will use to send the media. If there is a misconfiguration here, it could result in one-way or no audio.
  
  

• For example, if the IP address or port number in the "c=" line is incorrect, the call may not be able to transmit audio properly, leading to the issue described: users cannot hear the remote party.
  
  

3. Other SIP Fields and Their Roles:
   
   

• A. Contact: header: This header contains contact information for the user agent, specifically the URI where the endpoint is reachable. While this field is important for establishing the communication path, it does not directly provide information about the media flow, such as where audio will be sent or received.

• B. Allow: header: The "Allow" header indicates which methods are supported by the server. This helps with troubleshooting signaling issues but does not provide specific details about the media flow or potential issues related to audio transmission.

• C. o= line of SDP content: The "o=" line identifies the originator of the session and session information, such as the session ID and version. While this is essential for managing session control, it is not directly related to the media (audio) transmission itself. The "c=" line is more relevant in the context of media path troubleshooting.

• Given the symptoms, it is likely that the media stream is not being properly directed or negotiated, and examining the c= line of the SDP will provide insight into whether there is an issue with the IP address or port that is being used to send or receive audio.

• Troubleshooting should involve verifying that the media connection details (IP address and port) in the "c=" line match the actual network configuration and ensure that no firewall or NAT (Network Address Translation) issues are preventing the media from flowing correctly.

By checking and correcting any issues in the c= line of the SDP, you can resolve the problem where users are unable to hear each other during calls.

Question No 5:

What are some possible reasons for RTP traffic originating from an endpoint failing to be received by the destination endpoint?

RTP (Real-Time Transport Protocol) is commonly used in VoIP (Voice over IP) and video communications to deliver audio and video data in real-time. However, there are various reasons why RTP traffic may fail to reach its intended destination. Understanding these reasons is crucial for troubleshooting issues related to media transmission in such systems. Here are several potential causes:

A. The far-end connection data (c=) in the SDP was overwritten by deep packet inspection in the call signaling path.

In the context of VoIP calls, the Session Description Protocol (SDP) is responsible for defining the media parameters, such as IP addresses and ports, for the communication session. The c= field in SDP indicates the connection information, including the IP address and port for receiving RTP traffic. If deep packet inspection (DPI) or middleboxes (like firewalls or NAT devices) inspect the signaling traffic and modify the SDP information, it could inadvertently alter the c= field. This could result in RTP packets being directed to the wrong destination, causing the receiving endpoint to fail to get the media.

B. Cisco Unified Communications Manager invoked media termination point resources.

In some cases, a Media Termination Point (MTP) is used in VoIP networks to terminate media streams, allowing for specific media processing such as encryption, transcoding, or packetization. The Cisco Unified Communications Manager (CUCM) might invoke MTP resources during a call setup, which could redirect or terminate the RTP stream for reasons such as codec mismatch or network issues. If MTP resources are invoked, the media path may change, causing the RTP traffic to be sent to an unintended location, resulting in failure at the far endpoint.

C. The RTP traffic is arriving beyond the jitter buffer on the receiving end.

The jitter buffer is a mechanism at the receiving endpoint used to compensate for network delays and variations in packet arrival times. When packets arrive out of order or with excessive delay, they are temporarily held in the jitter buffer before being processed. If RTP packets arrive too late or in bursts that exceed the buffer's capacity, the packets might be discarded, leading to media loss. While this can affect call quality, it typically does not result in a complete failure to receive RTP traffic, but rather results in degraded quality.

D. A firewall in the media path is blocking TCP ports 16384-32768.

RTP typically uses UDP (not TCP) as its transport protocol, but firewalls may block UDP ports associated with RTP traffic. The range of UDP ports used for RTP is typically from 16384 to 32768. If a firewall in the media path is configured to block this range of ports, the RTP traffic will be blocked, preventing it from reaching the destination endpoint. This is a common issue in network configurations where firewalls are not properly configured to allow RTP traffic.

Answer:

The correct answer is D: A firewall in the media path is blocking TCP ports 16384-32768.

Explanation:

RTP traffic is critical for real-time communication, such as voice and video calls in VoIP applications. However, there are several potential reasons why RTP packets might not be received on the far endpoint.

1. Option A: Deep packet inspection (DPI) can be problematic if it modifies the SDP information in the signaling path. While this could cause the RTP traffic to be misdirected, it is generally less common compared to network issues like firewalls.

2. Option B: The Cisco Unified Communications Manager (CUCM) using MTP resources could certainly modify the media path, causing RTP traffic to be redirected. However, this scenario typically occurs under specific circumstances like media gateway interworking or transcoding, making it less likely to be the root cause.

3. Option C: Jitter buffer issues can result in packet loss or degraded call quality, but they usually don’t completely prevent RTP traffic from being received. It typically affects call quality rather than causing total failure.

4. Option D: A firewall blocking the RTP port range (16384-32768) is one of the most common and easily overlooked causes of RTP traffic failure. Many firewalls are configured to block UDP traffic, including the range used for RTP, either for security reasons or due to misconfiguration. When these ports are blocked, the RTP packets simply cannot reach the far endpoint, resulting in a complete failure of the media stream.

In conclusion, Option D is the most likely cause of RTP traffic failure, as improperly configured firewalls blocking the UDP port range used by RTP can easily prevent the media stream from reaching the destination endpoint, causing the communication to fail.

Question No 6:

An administrator is troubleshooting call failures on an H.323 gateway via the command-line interface (CLI). To capture and view the signaling related to both media and call setup during the troubleshooting process, 

Which debug command should the administrator enable to get the required information?

A. debug H.323 messages
B. debug H.225 asn1
C. debug H.246 asn1
D. debug H.225 media
E. debug H.323 asn1

Answer:

The correct answer is A. debug H.323 messages.

Explanation:

When dealing with call failures in H.323 environments, administrators need to gather information about the signaling messages exchanged between endpoints and gateways. H.323 is a signaling protocol used for voice and video communication, and it often integrates with other protocols such as H.225 and H.245 for call signaling and media control.

H.323 Signaling Overview:

H.323 is an umbrella protocol that covers various aspects of multimedia communications, including call control and media transport. The signaling part of an H.323 call setup involves multiple message exchanges, including setup, alerting, and teardown. The media control (for the actual voice or video communication) is controlled using H.245. Additionally, H.225 handles the call signaling, including call setup and teardown.

Breakdown of Each Option:

• A. debug H.323 messages: This command enables the debugging of all H.323 messages, including both signaling and media messages related to call setup and teardown. The administrator will get a comprehensive view of the signaling involved in the entire call process, which makes it the most appropriate command to troubleshoot call failures. It provides both higher-level signaling information and lower-level protocol messages, which are critical for understanding where the failure occurs during the setup or media exchange phases of a call.

• B. debug H.225 asn1: H.225 is a part of the H.323 protocol suite, responsible for the call signaling, such as call setup, call teardown, and admission control. The ASN.1 (Abstract Syntax Notation One) format is used for encoding messages. While this debug command provides detailed information on H.225 message encoding, it focuses primarily on call signaling and does not offer as complete a picture of the entire H.323 signaling, including media control messages.

• C. debug H.246 asn1: H.246 is not directly relevant to H.323 signaling. It is a part of the H.324 protocol, which is used for multimedia communications over circuit-switched networks. This is not applicable to the H.323 gateway troubleshooting scenario.

• D. debug H.225 media: H.225 also handles the call signaling part of H.323, but this specific command is focused only on media-related signaling within the H.225 context. It does not provide a full picture of the signaling across both call setup and media setup, and might be limited to the media-related control, which is not sufficient for troubleshooting overall call failures.

• E. debug H.323 asn1: Similar to option B, this command helps with debugging the H.323 signaling but focuses more on ASN.1 encoding and decoding issues rather than providing a broad overview of both signaling and media aspects of the call. It is more specific and less useful for comprehensive troubleshooting.

To troubleshoot call failures effectively, the administrator should enable debug H.323 messages as it provides a detailed and holistic view of both signaling and media setup. This ensures that the administrator can identify problems in either the signaling (such as call setup or teardown issues) or media setup (such as issues with media channels). By enabling this debug command, the administrator can capture the necessary messages to diagnose the issue more efficiently and resolve the problem.

Question No 7:

What is the first preference condition that is matched in a SIP-enabled incoming dial peer when handling an incoming call?

Options:

A. Incoming URI
B. Target carrier-ID
C. Answer-address
D. Incoming called-number

Correct Answer: D. Incoming called-number

Explanation:

In a Session Initiation Protocol (SIP)-enabled network, when a call comes in, the router or device needs to route that call to the appropriate destination based on certain parameters. These parameters are defined in incoming dial peers that guide the routing of incoming calls. The incoming dial peer in SIP is responsible for determining how an incoming call should be handled, based on various conditions such as the called number, carrier ID, and other criteria specified in the dial peer configuration.

The incoming dial peer is evaluated in the order of the most specific match to the least specific. This means that the router first tries to match the most specific condition in the dial peer before falling back to less specific conditions. The first preference condition that is matched plays a crucial role in how the router processes and routes the incoming call.

Let's look at each option in more detail:

1. Incoming called-number (Option D)
   This is the first preference condition when a SIP-enabled incoming dial peer is evaluated. The incoming called number refers to the destination number dialed by the caller. When an incoming call is received, the router or device first tries to match the called number specified in the dial peer configuration. If a match is found, this dial peer is selected for further processing. The called number is often the most specific and reliable parameter to use in call routing because it directly correlates with the destination of the call.

2. Incoming URI (Option A)
   The incoming URI is another parameter that can be used to match an incoming dial peer, but it is evaluated after the called number. The URI typically contains the address of the SIP service, and while it is useful for certain types of calls (like SIP-based routing), it is not the primary condition used for matching an incoming dial peer in most cases. The URI is more commonly used for outbound routing in SIP networks rather than incoming routing.

3. Target carrier-ID (Option B)
   The target carrier-ID specifies the identifier of a particular carrier or route. While important for determining the carrier to use for an outgoing call, it is not the first condition matched in an incoming call situation. The carrier ID may be used if there is no match on the called number or URI, but it is secondary to the called number in terms of preference.

4. Answer-address (Option C)
   The answer-address is typically used to define the address to which a call should be routed after it has been matched to a dial peer. It plays a secondary role in the process of call setup and routing. It is not the first preference condition and is usually considered after other parameters like the called number have been checked.

Why "Incoming Called-Number" is the First Preference:

The incoming called-number is generally the most accurate and straightforward way to match a dial peer because it directly reflects the number dialed by the user. In SIP call routing, it is common practice to match this field first because it gives an immediate and precise identifier for where the call should be routed.

Once the incoming called-number is matched, the router can then look at other parameters (like URI, carrier ID, and answer address) to refine the routing decision if needed. However, since the called number is typically a primary determinant of the destination in most SIP environments, it is the first condition that is evaluated when an incoming call is received.

When a SIP-enabled incoming dial peer processes an incoming call, the first preference condition it looks at is the incoming called-number. This is the most direct and reliable piece of information that determines the destination of the call. Other conditions like incoming URI, target carrier-ID, and answer-address may be used if the called-number does not match, but they are evaluated in a secondary order. This process ensures that calls are routed efficiently and correctly based on the destination number dialed by the caller.

Question No 8:

Cisco SIP-based IP telephony has been implemented across two floors in your company. However, users are reporting intermittent voice issues during calls between the two floors. These calls are being established correctly, but sometimes they experience one-way audio or no audio at all. After investigating, you discover that a firewall exists between the floors, and the firewall administrator confirms that it allows SIP signaling and UDP ports in the range of 20000 to 22000 bidirectionally. Given these conditions, 

What are two possible solutions to resolve the voice issue? (Choose two.)

A. Modify the SIP profile assigned to the IP phones in Cisco Unified Communications Manager (CUCM) to change the range of media ports to 16384-32767.
B. Ask the firewall administrator to change the allowed ports to TCP.
C. Ask the firewall administrator to change the range of UDP ports to 16384-32767.
D. Modify the SIP profile assigned to the IP phones in CUCM to change the range of media ports to 20000-22000.
E. Change the range of media ports to 20000-22000 in the System Parameters settings in Cisco Unified Communications Manager.

Answer:

The two most appropriate solutions are:

A. Modify the SIP profile assigned to the IP phones in Cisco Unified Communications Manager (CUCM) to change the range of media ports to 16384-32767.
C. Ask the firewall administrator to change the range of UDP ports to 16384-32767.

Explanation:

This scenario involves SIP-based telephony over a network with a firewall between the two floors. The key issue here is the one-way or no audio experienced during calls, which is often a result of improper handling of media ports by the firewall or CUCM settings.

1. SIP Media Ports: By default, Cisco IP phones use a range of ports for RTP (Real-time Transport Protocol) traffic to handle the actual voice stream during a call. The firewall between the floors needs to allow these RTP ports to pass bidirectionally. The default range for these RTP ports in many Cisco systems is 16384-32767. The firewall may be permitting ports 20000-22000, but this is too narrow, and it could block some of the media traffic necessary for proper communication.
   
   

• Solution A: You can modify the SIP profile in Cisco Unified Communications Manager (CUCM) to change the media port range to 16384-32767. This will ensure that all the necessary RTP traffic can traverse the firewall and ensure full-duplex (bidirectional) audio during calls.
  
  

• Solution C: The firewall should be configured to allow the full range of UDP ports from 16384 to 32767, which corresponds to the RTP media traffic. This is essential for proper communication between the two floors. Restricting the firewall to only 20000-22000 might result in blocked media streams and one-way or no audio during calls.
  
  

2. Why not the other options?
   
   

• Option B (Change ports to TCP): SIP signaling typically uses UDP (User Datagram Protocol) for low-latency, real-time communication. Changing to TCP would not resolve the issue, as SIP over TCP is not commonly used for voice traffic due to its higher overhead and potential latency.

• Option D (Change SIP profile to 20000-22000): This is the current configuration on the firewall, which is causing the issue. Changing the SIP profile to use this narrow port range would not fix the issue and might even exacerbate it.

• Option E (Change range in System Parameters in CUCM): This is not the best solution because it would apply globally to all media traffic, which could lead to other unintended network issues. It’s more precise and effective to change the media port range in the SIP profile for the affected phones.

In summary, to resolve the intermittent voice issues, the firewall should allow the broader RTP media port range of 16384-32767, and the CUCM settings should reflect this range to ensure proper media transmission.