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Last Updated: Oct 11, 2026
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Cisco 300-815 Exam Overview:
| Certification Vendor: | Cisco |
|---|---|
| Exam Name: | Implementing Cisco Advanced Call Control On-Premises |
| Exam Number: | 300-815 |
| Available Languages: | English |
| Real Exam Qty: | 55-65 |
| Certificate Validity Period: | 3 years |
| Exam Price: | $300 USD |
| Exam Format: | Scenario-based, Multiple response, Multiple choice, Drag and drop |
| Related Certifications: | Cisco Certified Specialist - Collaboration Call Control & Mobility Implementation CCNP Collaboration |
| Passing Score: | Scaled scoring (variable by exam version) |
| Exam Duration: | 90 minutes |
| Sample Questions: | DOWNLOAD DEMO |
| Exam Way: | Pearson VUE testing centers and online proctored delivery |
| Pre Condition: | No formal prerequisites; CCNP Collaboration requires passing 350-801 CLCOR and one concentration exam (300-815 qualifies) |
| Official Syllabus URL: | https://learningcontent.cisco.com/documents/marketing/exam-topics/300-815-CLACC-v2.0.pdf |
Cisco 300-815 Exam Syllabus Topics:
| Section | Weight | Objectives |
|---|---|---|
| Session Border Controller and Voice Gateway Technologies | 30% | - Configure Cisco UCME and SIP SRST - Configure advanced SIP interoperability with Cisco Unified Border Element
|
| Signaling and Media Protocols | 10% | - Troubleshoot advanced elements of a SIP conversation
|
| Mobility | 20% | - Configure and troubleshoot Cisco Unified Communications Manager Express and Cisco Unified Communications Manager mobility features
|
| Cisco Unified CM Call Control Features | 20% | - Configure Cisco Unified CM mobility features
|
| Call Control and Dial Planning | 20% | - Troubleshoot globalized call routing elements in Cisco Unified Communications Manager
|
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The Cisco Implementing Cisco Advanced Call Control and Mobility Services blueprint centers on these domains:
- Session Border Controller and Voice Gateway Technologies (30%)
- Signaling and Media Protocols (10%)
- Cisco Unified CM Call Control Features (20%)
Additional domains complete the official outline, and our bank covers every one.
By daily checking and automatic delivery. Our Cisco experts check for updates every day, and whenever the 300-815 bank is revised, the system sends the latest version to you automatically — free for 365 days from purchase. The bank stays closely linked to the actual exam content, with expert-verified answers throughout. Choose among three formats: a printable PDF for easy notes, a Windows PC test engine that simulates the real exam environment, and an online engine that runs on any device and works offline after the first download. A free demo lets you experience all of it before buying.
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Currently, the 300-815 exam's passing score is Scaled scoring (variable by exam version) and the registration fee is $300 USD. These figures belong to Cisco and can be revised, so confirm them on the official site before scheduling.
Cisco states the following prerequisites for the Cisco Implementing Cisco Advanced Call Control and Mobility Services: No formal prerequisites; CCNP Collaboration requires passing 350-801 CLCOR and one concentration exam (300-815 qualifies).
For the authoritative details, see the official certification page.
According to the latest exam information, the 300-815 exam contains 55-65 questions and lasts 90 minutes minutes. Simulating those conditions at home with the PC test engine is excellent preparation for the real environment.
Cisco Implementing Cisco Advanced Call Control and Mobility Services Sample Questions:
When you troubleshoot H.323 call setup, which message informs you that the called party is being notified about the call?
- A. ALERTING
- B. RINGING
- C. CONNECT
- D. PROCEEDING
Correct Answer: A 🗳️
A customer reports audio quality issues between video endpoints in the HQ location in California and one of the branches in Texas. Which two actions must RTCP take to troubleshoot this issue? (Choose two.)
- A. Allow for VAD to be used for calls using the G7.29 codec, which reduces the usage of the WAN bandwidth and saves around 30% of bandwidth per call.
- B. Configure the rtcp keepalive command in Cisco Unified Border Element to generate reports, which can be reviewed using the debug voip rtcp packet command.
- C. Encrypt the media to stop rogue devices from replying and putting that traffic back on the WAN, which avoids any extra bandwidth and ensures the quality of the calls.
- D. Compress the headers of RTP traffic to lower the bandwidth consumption over the WAN. which allows more calls with less bandwidth consumed.
- E. Gather statistics on a media connection and information such as packets sent, lost packets, jitter, feedback, and round-trip delay. This information can help isolate the type of audio quality issues and the direction of the affected traffic.
Correct Answer: D,E 🗳️
A new solution is configured to support internal, local, and international calling. Calling [+44 1111 1111] from one of the registered internal phones does not work. Local and internal calls seem to work without any problems. The configuration has patterns configured to match the failing dialed number [+44 .... ....]. The other configured patterns show [2...] for internal numbers and [555 ...] for local numbers. International numbers use E.164 as recommended. What is missing to make this solution work?
- A. \ in front of the +
- B. =+ cannot be used in a route pattern, only in a SIP pattern
- C. / in front of the +
- D. 001 or 00 must be used instead of the + sign on Cisco UCM
Correct Answer: A 🗳️
Refer to the exhibit.
An engineer is troubleshooting a call-establishment problem between Cisco Unified Border Element and Cisco UCM. Which command set corrects the issue?
- A. SIP binding In SIP configuration mode:
voice service volp
sip
bind control source-Interface GlgabltEthernetO/0/1 bind media source-Interface GlgabltEthernetO/0/1 - B. SIP binding in SIP configuration mode:
voice service voip sip
bind control source-interface GigabitEthernetO/0/0 bind media source-interface GigabitEthernetO/0/0 - C. SIP binding in dial-peer configuration mode:
dial-peer voice 100 volp
voice-class sip bind control source-interface GigabitEthernetO/0/0
voice-class sip bind media source-interface GigabitEthernetO/0/0 - D. SIP binding In dial-peer configuration mode:
dial-peer voice 300 voip
voice-class sip bind control source-interface GigabitEthernetO/0/1 voice-class sip bind media source-interface GigabitEthernetO/0/1
Correct Answer: C 🗳️
An engineer Is configuring a Cisco Unified Border Element. The SIP ITSP requires a DNIS of 10-digits and DTMF tones sent in the RTP stream on the outbound leg of a call. Which code completes this configuration?
- A.
- B.
- C.
- D.
Correct Answer: D 🗳️
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