Businesses need cloud-connected audio to operate in today’s global environment. Remote work trends have pushed organizations to adapt quickly. Statistics show that U.S. companies make up 58% of Cisco WebEx Cloud Connected Audio’s customer base. Large enterprises with over 1000 employees represent 51% of these customers.
Cloud-connected audio blends audio conferencing solutions with cloud-based collaboration platforms. Cisco WebEx Cloud Connected Audio creates a dedicated IP connection that links Service Provider’s data center directly to a Cisco Webex data center. This setup offers a more reliable audio conferencing experience than traditional methods. Users can quickly rejoin their meetings if audio problems occur.
This piece guides engineers through cloud-connected audio system implementation. They will learn about core architecture, deployment, integration, and troubleshooting steps. IT professionals can build reliable cloud audio conferencing capabilities for their organizations with this knowledge.
Understanding Cloud Connected Audio Architecture
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Cloud connected audio fundamentally changes how enterprise audio conferencing works. The architecture brings audio, video, and web collaboration together on a unified platform through cloud technology. Users can communicate smoothly no matter where they are or what device they use.
What is Cloud Connected Audio in Enterprise Environments?
Cloud connected audio creates a direct connection between an organization’s communication system and cloud-based collaboration platforms. Companies can extend their existing IP telephony investments while adding integrated audio features. Businesses typically choose between two deployment options: getting it as-a-service from a provider or building their own enterprise architecture.
A direct enterprise setup connects the organization’s network straight to the collaboration cloud. This removes geographic and tech barriers that often make remote meetings difficult. Businesses gain several key benefits:
- Monthly costs they can predict without overages
- No PSTN transport costs for on-net users
- Easier IT support and maintenance
- Better scaling without worrying about conference size limits
The system works great with existing Unified Communications infrastructure. Research shows that 84% of enterprises report better employee productivity when they use collaboration technology.
Cloud Connected Meaning in Audio Conferencing
“Cloud connected” in audio conferencing means traditional audio systems integrate with cloud-based platforms through dedicated IP connections. This creates a direct path for audio traffic between company networks and cloud provider data centers.
Webex Cloud Connected Audio sets up direct IP peering between the service provider’s data center and a Cisco Webex data center. Audio conference calls from both off-net and on-net users flow through the provider’s phone network. Cisco handles the audio mixing.
The system creates VoIP paths for registered devices, which makes call routing simpler. Companies can still use their PSTN services for non-registered phones or mobile devices. This hybrid setup makes the most of existing infrastructure while adding cloud features.
Cisco Webex Cloud Connected Audio vs Traditional PSTN
Traditional PSTN audio conferencing has served businesses well for decades with its circuit-switched technology. Cisco Webex Cloud Connected Audio offers many advantages over this older approach.
Traditional PSTN charges for each call to conference bridges. Webex Cloud Connected Audio gives you predictable monthly costs without overages. Industry analysts predicted that by 2021, 60% of web conferences would use embedded audio instead of separate PSTN connections.
These systems have major architectural differences:
| Feature | Traditional PSTN | Cisco Webex Cloud Connected Audio |
| Connection Type | Circuit-switched telephony | Direct IP peering with data centers |
| Cost Structure | Per-minute charges, potential overages | Predictable monthly costs |
| Audio Quality | Subject to PSTN limitations | Enhanced with wide-band codecs |
| Integration | Separate systems for audio and web | Natively integrated experience |
| Scalability | May require additional hardware | Cloud-based scaling without infrastructure changes |
Webex Cloud Connected Audio delivers consistent experiences across all devices. Internal and external users can work together easily from anywhere. Audio calls route through the internet or dedicated Edge Connect circuits to the Webex cloud. This approach separates PSTN from Webex by routing calls through the customer’s on-premises UC system.
Companies can choose between internet connections or dedicated direct peering links (Edge Connect) to Webex. This flexibility lets them match their needs and current infrastructure. The architecture includes toll fraud protection to block unauthorized access to the phone system.
To wrap up, a solid grasp of cloud connected audio architecture helps companies implement these systems well. The next section explores the core components needed to build a working cloud connected audio system.
Core Components of a Cloud Connected Audio System
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Cloud-connected audio systems need several components that work together. These technical elements are the foundations of the solution that blends on-premises infrastructure with cloud services.
Audio Bridge Integration with Cisco Collaboration Cloud
The audio bridge mixes conference audio in a cloud-connected audio system. Cisco Webex Cloud Connected Audio lets Cisco handle audio mixing directly in their collaboration cloud. This setup creates a natural path where audio and web collaboration exist together in the same framework.
The Service Provider Peering Architecture connects the Service Provider’s data center to a Cisco Webex data center through direct IP. All audio conferencing calls flow over the service provider’s telephony network while Cisco handles the audio mixing. This setup changes the traditional conferencing workflow by:
- Getting rid of separate systems for audio and web collaboration
- Making user experience consistent across devices
- Delivering natively through the Cisco Collaboration Cloud
The audio bridge integration lets users choose flexible calling options like dial-in, dial-out, call back, integrated VoIP, and hybrid audio setups. Users get the same high-quality audio no matter where they are or how they connect.
Direct IP Peering with Cisco Collaboration Cloud
Direct IP peering connects an organization’s network to Cisco Webex data centers. Webex Edge Connect manages this connection through a dedicated QoS-enabled peering link that protects meetings from public internet issues.
Organizations typically set up equipment in two data centers for access and backup. Each location needs:
- Carrier Ethernet circuits that connect to a Service Provider backbone network
- Edge routers that manage the network
- Cisco Unified Border Elements (CUBEs) that control sessions
- DSP farms that process media
Organizations using Webex Edge Connect connect through a cloud exchange provider (Equinix or Megaport) to link their routing edge to Webex backbone edge routers. This setup offers key benefits:
- Less network latency, packet loss, and jitter
- Guaranteed bandwidth with direct connectivity
- Better security against internet threats
Edge Connect peering needs an active connection on the Equinix or Megaport Cloud Exchange, a BGP Autonomous System Number, customer IP addresses for BGP peering, and network devices that run BGP and 802.1Q tagging.
VoIP and PSTN Interoperability in Hybrid Deployments
Some organizations can’t switch to VoIP right away because of existing investments, geographic limits, or regulations. Hybrid deployments blend VoIP and PSTN connectivity to offer a middle ground.
Webex Edge Audio creates an end-to-end VoIP path for devices registered with Cisco Unified Communications Manager in hybrid setups. Non-registered phones or mobile devices can use the organization’s PSTN services. This approach offers several benefits:
- Separating PSTN from Webex by routing calls through an on-network path
- Letting participants join meetings through direct VoIP routes or existing PSTN services
- Keeping current investments while adding cloud features
- Choosing the most cost-effective path based on location
Cisco Unified Border Element (CUBE) works as an enterprise-class Session Border Controller to implement this setup. CUBE connects enterprise networks with cloud collaboration services. It handles signaling and media traffic while creating a secure boundary between internal and external services.
Hybrid deployments work best for organizations that have invested heavily in legacy telephony infrastructure or face geographic bandwidth limitations. They can start using cloud-connected audio right away while moving away from PSTN gradually.
Network and Security Requirements for Implementation
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Cloud-connected audio implementation security needs close attention to network setup and security protocols. Organizations that use Cisco Webex Cloud Connected Audio should set up proper security measures. These measures protect both signaling and media traffic while meeting industry standards.
Firewall and NAT Configuration for Audio Traffic
Network Address Translation (NAT) and firewall configuration are the foundations of cloud-connected audio systems. Most enterprise networks use private IP addressing schemes, which makes NAT traversal vital. SIP-based VoIP solutions face a technical challenge to work correctly when firewalls and NAT devices exist between endpoints that exchange data.
Cisco ASA firewalls version 8.3 and later check interface Access Control Lists (ACLs) after untranslating packets. This basic change affects how the core team must set up their environments. These systems offer two types of NAT:
- Object NAT (Auto NAT): Set up within network object definitions
- Manual NAT (Twice NAT): Gives more detailed control but needs careful rule ordering
Firewalls come with default security behaviors that you should know before applying ACLs to interfaces:
- Traffic from lower to higher security interfaces is denied
- Traffic from higher to lower security interfaces is allowed
Engineers should use real IP addresses of hosts instead of translated IPs and explicitly allow required traffic flows when setting up ACLs. UDP-based applications like cloud-connected audio work better with UDP timeout set to 120 seconds because SIP sends keep-alive messages about every 25 seconds.
TLS Encryption and SRTP for Secure Media Transport
Secure communication protocols are vital to cloud-connected audio security. Transport Layer Security (TLS) encrypts SIP signaling, while Secure Real-time Transport Protocol (SRTP) protects the actual media content.
TLS versions 1.2 and 1.3 support Cloud-Connected UC and create encrypted connections between endpoints. SRTP adds encryption, confidentiality assurance, and message authentication to protect WebRTC conversations without hurting service quality.
Webex apps and Cisco Video devices use these encryption ciphers:
- AES-256-GCM (preferred)
- AES-CM-128-HMAC-SHA1-80
Cloud-connected audio systems use transport protocols in this order:
- UDP (ports 5004 and 9000)
- TCP (port 5004) – fallback when UDP is blocked
- TLS (port 443) – final fallback option
TLS/SRTP provides end-to-end security between communicating endpoints. The DTLS handshake sets up keying material and algorithms for SRTP, which ensures device-to-device security without depending only on signaling security.
Compliance with ISO/IEC 27001 and SOC 2 Standards
Enterprise cloud-connected audio implementations must follow industry compliance standards. ISO/IEC 27001 and SOC 2 are two main frameworks for cloud services security.
ISO/IEC 27001:2022 describes what an information security management system (ISMS) needs and lists best practices and security controls for managing information risks. This standard doesn’t require specific controls but offers a detailed framework for security management. Organizations can get Cisco’s ISO/IEC 27001 certificates through the Compliance Reports Manager.
SOC 2 reports come from objective third parties who verify controls that protect customer data. These reports have two types:
- Type I: Looks at control design at a specific point
- Type II: Reviews design and effectiveness over time (Cisco only issues Type II reports)
Session Border Controllers (SBCs) help maintain security compliance. They check all voice/video traffic thoroughly and authenticate packets while blocking unauthorized ones. Well-configured SBCs can block all SIP packets except those with TLS protection, which safeguards enterprise voice communications.
Organizations should protect all transmitted configuration files, signaling, and media streams in their cloud-connected audio systems. Unprotected communications face the risk of interception and potential alteration.
Step-by-Step Deployment Guide for Engineers
Cisco Webex Cloud Connected Audio solution implementation needs careful planning and execution. The engineering team must follow specific deployment steps that will give a smooth integration between on-premises infrastructure and the Webex cloud.
Provisioning Cisco Webex Cloud Connected Audio
The engineering team starts with the provisioning process for Cloud Connected Audio. The process begins by generating a Letter of Authorization (LOA) for each peering circuit. Service providers need this LOA which contains critical network peering information. Engineers should complete these steps after circuit installation:
- Enter network peering information including ASN numbers, technical contact details, and peering IP addresses
- Configure BGP provisioning parameters including BFD intervals if required
- Submit network information to Cisco for approval
- Enter telephony information for Session Border Controllers (SBCs), Cisco Unified SIP Proxy (CUSP), and DNIS configurations
Cloud-based deployments need Cloud Connected PSTN as the connection type for locations before choosing the appropriate CCP provider. The final step specifies whether customers need WebEx audio options alongside Cloud Connected Audio.
Configuring SIP Trunks and Dial Plans
SIP trunk setup forms the foundations of connecting Cisco Unified Communications Manager to external devices during configuration. The proper SIP trunk configuration includes:
- Create a SIP trunk security profile with appropriate authentication and encryption settings
- Configure Common Device Configuration for IP addressing priorities
- In the Trunk Configuration window, assign the SIP profile and security profile to your SIP trunk
- Configure destination addresses using IPv4/IPv6, domain names, or DNS SRV records
Dial plans route calls to on-premises destinations through trunks or route groups. These plans specify routing choices for calls that match specific dial patterns. The dial plan configuration follows trunk and route group setup:
- Navigate to the Dial Plans section in Control Hub
- Create a named dial plan and select the appropriate routing choice
- Manually create or import dial patterns for selected route choices
Testing Audio Quality and Failover Scenarios
System reliability depends on rigorous testing. Failover testing shows if software can handle additional resources and maintain backup processes during failures. A comprehensive failover testing approach includes:
- Create a detailed test plan that thinks about budget, time, and resources
- Determine potential error conditions and analyze mechanisms
- Develop test cases for both scheduled and unplanned failover scenarios
- Execute tests in a controlled environment and measure recovery times
Cisco Control Hub provides immediate monitoring for audio quality testing. Failover recording servers check recording server states every 0.5 seconds and mark them unavailable after 2 seconds without response. Cold standby failover servers become operational in about five seconds plus connection time. Hot standby servers activate faster.
Integration with Existing Collaboration Tools
Cloud connected audio deployments work best when they integrate smoothly with your workplace tools. Cisco Webex Cloud Connected Audio shows its real value through collaboration with enterprise collaboration platforms.
Connecting Webex Cloud Connected Audio with Microsoft Teams
Engineers can choose from several ways to integrate Cisco Webex Cloud Connected Audio with Microsoft Teams. The main approach uses the Cisco Video Integration for Microsoft Teams (VIMT). This solution comes with better features like participant lists, participant mute controls, and two-way content sharing. WebRTC serves as a backup integration path when VIMT isn’t an option.
Administrators can set up automatic meeting joins through the Hybrid Calendar Connector. This lets users take advantage of the One Button to Push (OBTP) feature. Users can join any Microsoft Teams meeting with just one button press, no matter which tenant hosts the meeting.
The year 2023 brought a new option for organizations to run native Microsoft Teams Rooms on certified Cisco devices. These devices work well in spaces of all sizes, from small huddle rooms to large conference areas.
API-Based Integration with Enterprise Scheduling Systems
API integration connects cloud audio systems with enterprise scheduling platforms. This automation simplifies processes and makes employees happier. The scheduling API helps developers manage calendar data automatically. They can create, read, update, and delete events without manual work.
Enterprise deployments don’t need custom connections for each service provider like Google Calendar and Microsoft Outlook Calendar. Pre-built integrations handle all the communication between apps and calendar services.
Multi-device Support: Desk Phones, Soft Clients, and Mobile
Cloud connected audio solutions shine in supporting different types of devices. Companies can create consistent audio experiences on desk phones, softphone apps, and mobile clients.
Two-way presence sync lets users see if their contacts are in Webex calls, meetings, or using Do Not Disturb. Microsoft Teams and Webex status updates happen automatically, creating one unified presence system.
Cisco devices work great with different platforms. Users can join meetings from Webex, Microsoft Teams, Google Meet, and Zoom. The hardware provides high-quality audio no matter which meeting platform they choose. Recent updates allow content sharing through HDMI connections between Google Meet and Cisco Webex devices.
Monitoring, Troubleshooting, and Optimization
Cloud-connected audio environments need reliable monitoring tools, troubleshooting protocols, and network optimization techniques to maintain peak performance. Reliable operations depend on effective management after deployment.
Using Cisco Control Hub for Real-Time Monitoring
Cisco Control Hub gives complete visibility into Webex services performance through immediate analytics and diagnostic features. The central management platform updates data every minute. This helps administrators spot and fix issues quickly. IT teams can access ongoing meetings right away through the Live Monitoring Dashboard without searching. This lets them fix problems before users notice them.
Control Hub’s troubleshooting tools show detailed meeting insights for each participant with significant metrics like:
- Join and exit times for each participant
- Audio and video quality measurements
- Content sharing analytics
- Packet loss, latency, and jitter statistics
- CPU usage data
The Admin Join feature lets IT administrators experience user issues during live meetings. This gives them a direct view of quality problems.
Common Audio Issues: Echo, Latency, and Packet Loss
Three main technical issues affect cloud connected audio quality:
Echo happens mostly from speaker-microphone feedback loops. This occurs when a microphone picks up audio from speakers and sends it back into the call. Echo cancelation features can fix this issue when configured correctly.
Latency shows up as noticeable delays over 250ms between participants. People end up talking over each other and conversations become awkward. Network congestion, low bandwidth, and too many internet hops cause this issue.
Packet loss creates choppy or dropped audio segments that make conversations hard to follow. Teams should investigate right away when packet loss goes above 1% in audio conference calls. Forward Error Correction (FEC) helps reduce this by sending extra data packets to rebuild missing information.
QoS Configuration for Prioritizing Audio Traffic
Quality of Service settings give priority to time-sensitive network traffic. Good QoS configurations use Differentiated Services Code Point (DSCP) markings with port-based Access Control Lists to identify and sort packets.
Audio performs best with this QoS priority order:
- PTPv2 clock synchronization (highest priority)
- Audio streams (second highest)
- Video streams (third highest)
- Control/management traffic (lowest priority)
Networks without proper QoS may face jitter, packet loss, and delayed round-trip time. These issues lead to garbled audio, missing words, and noticeable conversation delays. Another important factor for distributed audio systems especially for hybrid meetings or recorded sessions is keeping the sound clean and clear. Tools like a vocal remover can help engineers or editors separate voice tracks from background music or remove vocals completely. This makes it easier to reuse background sounds, add new voice-overs, or fix audio before uploading it. Doing this kind of basic audio cleanup helps maintain professional sound quality in both live and recorded content
High-bandwidth networks might work fine without QoS when usage stays under 50%. However, busy networks need these settings.
Organizations can eliminate delays and packet issues by adjusting port range sizes for different traffic types. This works well when all network devices use the same classifications and markings.
Conclusion
Cloud connected audio has without doubt changed how enterprises communicate by blending traditional audio conferencing with cloud-based collaboration platforms. This piece explores the detailed architecture, components, and ways to implement successful deployment strategies.
The foundation for implementation starts with a clear grasp of architectural differences between traditional PSTN and cloud connected audio. Direct IP peering with Webex data centers removes geographical barriers. It also provides better reliability at predictable costs. The core components work together to create a unified communication experience through audio bridge integration, direct IP peering, and hybrid VoIP/PSTN solutions.
Security plays a vital role in enterprise deployment. Data remains protected through proper firewall configuration, TLS encryption, and SRTP for media transport. These measures help meet compliance standards like ISO/IEC 27001 and SOC 2. Engineers should follow a systematic deployment process from original provisioning through SIP trunk configuration and thorough testing.
Knowing how to blend with existing collaboration tools adds more value to cloud connected audio. Microsoft Teams integration and API connections to scheduling systems create a unified ecosystem. Multi-device support prevents isolated communication channels. Cisco Control Hub’s immediate insights help monitor performance after deployment. Teams can fix common issues like echo, latency, and packet loss to maintain peak performance.
Organizations that use cloud connected audio ended up with better communication quality and lower costs. Management became simpler with improved scalability. This technology connects traditional telephony with modern cloud collaboration. Engineers now have a robust framework to build resilient, future-proof audio conferencing solutions that adapt to business needs.
FAQs
1. What is cloud connected audio and how does it differ from traditional audio conferencing?
Cloud connected audio integrates audio conferencing with cloud-based collaboration platforms through direct IP connections. Unlike traditional PSTN-based systems, it offers predictable costs, enhanced audio quality, and seamless integration with web conferencing tools.
2. What are the key components needed to implement a cloud connected audio system?
The core components include audio bridge integration with the collaboration cloud, direct IP peering with data centers, and VoIP/PSTN interoperability for hybrid deployments. These work together to create a unified communication experience.
3. How can engineers ensure security in cloud connected audio implementations?
Engineers should configure firewalls and NAT properly, implement TLS encryption for signaling and SRTP for media transport, and ensure compliance with standards like ISO/IEC 27001 and SOC 2. Using Session Border Controllers (SBCs) also helps enhance security.
4. Can cloud connected audio integrate with existing collaboration tools like Microsoft Teams?
Yes, cloud connected audio can integrate with tools like Microsoft Teams through options such as Cisco Video Integration for Microsoft Teams (VIMT) or WebRTC. It can also connect with enterprise scheduling systems via APIs and supports multiple devices including desk phones, soft clients, and mobile devices.
5. What are some common audio issues in cloud connected systems and how can they be addressed?
Common issues include echo, latency, and packet loss. These can be addressed through proper echo cancelation, network optimization to reduce latency, and implementing Quality of Service (QoS) configurations to prioritize audio traffic and minimize packet loss.