Telephony Architecture¶

Overview¶

This document details the telephony infrastructure design for Webex Contact Center, including PSTN connectivity, call routing, number management, and voice quality optimization.

1. PSTN Connectivity Options¶

1.1 Design Decision: On-Premises Cisco CUBE (Selected for This Migration)¶

Based on the discovery phase and CUBE design chapter findings, on-premises Cisco CUBE has been selected as the primary PSTN connectivity option for this Avaya to Webex Contact Center migration.

Why On-Premises CUBE Was Selected: - ✅ Leverage existing carrier contracts and SIP trunks - ✅ Retain control over routing and DID management - ✅ Keep existing DIDs without complex porting scenarios - ✅ Maintain premises-based control during migration phases - ✅ Support hybrid coexistence with Avaya during transition - ✅ Meet enterprise security and compliance requirements

1.2 Selected Architecture: On-Premises Cisco CUBE¶

┌─────────────────┐
│  PSTN Network   │
│ (Existing SIP   │
│    Provider)    │
└────────┬────────┘
         │
         │ SIP Trunk (Existing)
         │
┌────────▼────────┐
│  Cisco CUBE     │
│  (On-Premises)  │
│                 │
│ • ASR 1002-HX   │
│ • Dual HA Setup │
│ • TLS 1.2+ SIP  │
│ • SRTP Media    │
└────────┬────────┘
         │
         │ Secure SIP/TLS over Internet
         │
┌────────▼────────┐
│ Webex Contact   │
│ Center Cloud    │
│ (Datacenter)    │
└─────────────────┘

Key Components: - CUBE Hardware: Cisco ASR 1002-HX (2× units for HA) - SIP Trunks: Existing carrier relationship maintained - Connectivity: Dual ISP with 1 Gbps total bandwidth - Security: TLS 1.2+ for signaling, SRTP for media encryption - Registration: CUBE acts as Local Gateway to Webex Calling

Reference Documents: - See Chapter 2: CUBE & SBC Design for complete session sizing (6,084 sessions for 1,000 agents) - See Chapter 3: Network and Security for firewall rules and CUBE dial-peer configurations

1.3 Inbound Voice Call Flow (On-Premises CUBE)¶

┌─────────────────┐
│   Customer      │
│   Dials TFN     │
│ 1-800-XX5-0100  │
└────────┬────────┘
         │
         │ PSTN/SIP
         │
┌────────▼────────┐
│  SIP Provider   │
│ (Existing ITSP) │
└────────┬────────┘
         │
         │ SIP Trunk
         │ (Existing)
         │
┌────────▼────────────────────┐
│     CISCO CUBE              │
│     (On-Premises)           │
│                             │
│  ┌─────────────────────┐    │
│  │ Inbound Dial-Peer   │    │
│  │ • Match DNIS        │    │
│  │ • Apply Translation │    │
│  │ • Select Codec      │    │
│  │ • Route to WxCC     │    │
│  └──────────┬──────────┘    │
│             │               │
│  ┌──────────▼──────────┐    │
│  │ Security Processing │    │
│  │ • TLS Encryption    │    │
│  │ • SRTP Setup        │    │
│  │ • SIP Header Mods   │    │
│  └──────────┬──────────┘    │
│             │               │
└─────────────┼───────────────┘
              │
              │ SIP/TLS (Port 5061)
              │ + SRTP Media
              │
┌─────────────▼───────────────┐
│    WEBEX CONTACT CENTER     │
│    (Cloud Platform)         │
│                             │
│  ┌─────────────────────┐    │
│  │ Entry Point         │    │
│  │ • EP_Sales_TF       │    │
│  │ • Match DNIS        │    │
│  └──────────┬──────────┘    │
│             │               │
│  ┌──────────▼──────────┐    │
│  │ Flow Designer       │    │
│  │ • IVR Prompts       │    │
│  │ • Self-Service      │    │
│  │ • Queue Decision    │    │
│  └──────────┬──────────┘    │
│             │               │
│  ┌──────────▼──────────┐    │
│  │ Queue Routing       │    │
│  │ • Skills Match      │    │
│  │ • Agent Selection   │    │
│  │ • Priority Rules    │    │
│  └──────────┬──────────┘    │
│             │               │
└─────────────┼───────────────┘
              │
              │ WebRTC/Webex Media
              │
┌─────────────▼───────────────┐
│     AGENT DESKTOP           │
│     (Webex App)             │
│                             │
│  • Screen Pop (CRM Data)    │
│  • Call Controls            │
│  • Recording Active         │
└─────────────────────────────┘

Detailed Call Flow Steps:

Step	Component	Action
1	Customer	Dials 1-800-XX5-0100
2	PSTN Carrier	Routes call via SIP to CUBE
3	CUBE (Inbound)	Receives SIP INVITE, matches dial-peer
4	CUBE (Translation)	Applies number translation rules
5	CUBE (Security)	Encrypts with TLS/SRTP for Webex
6	CUBE (Outbound)	Sends to Webex CC cloud endpoints
7	Webex CC Entry Point	Matches DNIS to EP_Sales_TF
8	Flow Designer	Executes IVR logic (menu, self-service)
9	Queue Engine	Routes to Sales_Queue based on skills
10	Agent Desktop	Call delivered to available agent
11	CRM Integration	Screen pop with customer data
12	Recording	Call recording initiated

1.4 Alternative Option: Cisco VPOP (Cloud-Connected PSTN)¶

Note: This option was evaluated during discovery but NOT selected for this migration. Documented here for reference and future consideration.

Cisco VPOP (Virtual Point of Presence)

┌─────────────────┐
│  PSTN Network   │
│   (Carriers)    │
└────────┬────────┘
         │
         │ Traditional TDM/SIP
         │
┌────────▼────────┐
│  Cisco VPOP     │
│  (Cloud SBC)    │
│                 │
│ • SIP Gateway   │
│ • Transcoding   │
│ • Protocol Conv │
└────────┬────────┘
         │
         │ Secure SIP (TLS)
         │
┌────────▼────────┐
│ Webex Calling   │
│   Platform      │
└────────┬────────┘
         │
┌────────▼────────┐
│ Webex Contact   │
│    Center       │
└─────────────────┘

Benefits: - ✅ Managed by Cisco (no hardware to maintain) - ✅ Global carrier relationships - ✅ Automatic failover and redundancy - ✅ Built-in SBC and security - ✅ Simplified number porting - ✅ Quick deployment (2-4 weeks)

Why Not Selected for This Migration: - ❌ Requires porting all DIDs to Cisco-managed carriers - ❌ Lose existing carrier contracts and negotiated rates - ❌ Less control during phased migration - ❌ Monthly per-seat licensing adds cost - ❌ DID porting complexity for 265+ numbers

Future Consideration: VPOP may be reconsidered post-migration for: - Expansion to new regions - Simplified international connectivity - Reducing on-premises infrastructure

1.5 Hybrid Approach (Future State)¶

Primary Path (Current Design):
Customer Call ──► SIP Provider ──► On-Prem CUBE ──► Webex CC
(100% of calls during migration)

Potential Future Enhancement:
Primary: On-Prem CUBE (Existing DIDs)
Backup:  Cisco VPOP (New international numbers)

1.6 Comparison Matrix¶

Criteria	On-Prem CUBE (Selected)	Cisco VPOP (Alternative)
Control	Full premises control	Cisco-managed
Existing Carriers	✅ Keep existing contracts	❌ Port to Cisco carriers
DID Management	✅ No porting required	❌ Complex porting process
Migration Phases	✅ Supports hybrid coexistence	⚠️ All-or-nothing approach
Hardware	Customer-owned ASR/ISR	No hardware required
Expertise Required	High (CUBE configuration)	Low (Cisco-managed)
Cost Model	CapEx + existing OpEx	Monthly per-seat subscription
Deployment Time	4-8 weeks (with HA)	2-4 weeks
Security Control	Full policy control	Cisco-managed policies
Failover	Customer-configured HSRP	Automatic Cisco failover

Recommendation: On-Premises CUBE provides the best path for this enterprise migration due to existing infrastructure investments, carrier relationships, and phased migration requirements.

2. Number Management Strategy¶

2.1 Number Types and Allocation¶

Number Type	Use Case	Quantity	Management
Toll-Free	Primary customer contact	15	Ported to Webex
Local DID	Regional/office direct lines	250	Ported to Webex
International	Global customer access	30	New via Cisco VPOP
Internal Extensions	Agent/supervisor direct dial	500	Webex Calling
Test/Development	Testing environments	10	Temporary assignment

2.2 Number Porting Process¶

Phase 1: Pre-Port Planning (Weeks 1-2)

Action Items:
☐ Inventory all existing numbers
☐ Obtain Letters of Authorization (LOA)
☐ Identify port-in dates
☐ Plan for port validation testing
☐ Communicate customer-facing changes

Phase 2: Port Submission (Week 3)

Submit to Carrier:
• LOA (signed by authorized representative)
• Current carrier account information
• CSR (Customer Service Record)
• Desired port date/time
• Emergency callback information

Phase 3: Port Execution (Week 4)

Port Day Timeline:
T-24 hours: Final validation testing
T-4 hours:  Activate new routes in Webex
T-0:        Port executes at carrier
T+1 hour:   Validation testing
T+4 hours:  Full production traffic
T+24 hours: Post-port monitoring

Rollback Plan:

If port fails:
1. Immediately contact losing carrier
2. Request port cancellation
3. Restore original routing
4. Investigate and reschedule

3. Call Routing Architecture¶

3.1 Entry Point Design¶

Entry Point Configuration Matrix

Entry Point	DNIS	Purpose	Routing Destination
EP_Sales_TF	1-800-XX5-0100	Sales inquiries	Sales IVR Flow
EP_Support_TF	1-800-XX5-0200	Technical support	Support IVR Flow
EP_Billing	1-800-XX5-0300	Billing questions	Billing Queue Direct
EP_Spanish	1-800-XX5-0150	Spanish language	Spanish IVR Flow
EP_VIP	1-800-XX5-0500	Premium customers	VIP Queue Priority

Entry Point Routing Logic

┌──────────────────┐
│  Inbound Call    │
└────────┬─────────┘
         │
         ▼
┌──────────────────┐
│  DNIS Matching   │
│ (Number Dialed)  │
└────────┬─────────┘
         │
    ┌────┴────┐
    │         │
    ▼         ▼
┌────────┐ ┌────────┐
│Direct  │ │  IVR   │
│Queue   │ │ Flow   │
└────────┘ └───┬────┘
               │
         ┌─────┴─────┐
         │           │
    ┌────▼────┐ ┌────▼────┐
    │ Queue A │ │ Queue B │
    └─────────┘ └─────────┘

3.2 Queue Routing Strategies¶

Strategy 1: Longest Available Agent - Routes to agent idle for longest time - Ensures balanced distribution - Best for general queues

Strategy 2: Skills-Based with Proficiency - Matches call requirements to agent skills - Considers skill level (1-10) - Routes to highest skilled available agent

Strategy 3: VIP Priority Routing - Identifies high-value customers - Jumps queue position - Routes to specialized agents

Strategy 4: Business Hours Routing

IF current_time IN business_hours THEN
    Route to → Live Queue
ELSE IF current_time IN after_hours THEN
    Route to → Voicemail/Callback
ELSE
    Route to → Closed Message
END IF

3.3 Overflow and Failover¶

Overflow Routing Configuration

Primary Queue: Sales_Queue
├─ Service Level: Answer within 20 seconds
├─ Maximum Wait: 5 minutes
│
├─ Overflow Condition 1: Wait time > 3 minutes
│   └─ Action: Route to → General_Support_Queue
│
├─ Overflow Condition 2: Queue depth > 20 calls
│   └─ Action: Route to → Overflow_Queue
│
└─ Failover Condition: All agents logged out
    └─ Action: Play message → Offer callback

Geographic Failover

┌──────────────────┐
│   Primary Site   │
│   (US East)      │
│   └─ 60% agents  │
└────────┬─────────┘
         │
    Site failure?
         │
         ▼
┌──────────────────┐
│  Secondary Site  │
│   (US West)      │
│   └─ 40% agents  │
└──────────────────┘

4. Voice Quality Optimization¶

4.1 Codec Selection¶

Supported Codecs (Priority Order)

Codec	Bandwidth	Quality	Use Case
G.722	64 kbps	HD Voice	Internal, high bandwidth
G.711	64 kbps	Toll Quality	Most common, PSTN
G.729	8 kbps	Compressed	Low bandwidth scenarios
Opus	Variable	Adaptive	Future/optimal performance

Codec Negotiation Strategy:

Offer: G.722, G.711, G.729
Prefer: G.722 (HD voice when available)
Fallback: G.711 (universal compatibility)
Emergency: G.729 (bandwidth constraints)

4.2 QoS Configuration¶

DSCP Marking Standards

Traffic Type	DSCP Value	Priority	Bandwidth
Voice (RTP)	EF (46)	Highest	100 kbps/call
Signaling (SIP)	CS3 (24)	High	10 kbps/call
Video	AF41 (34)	Medium-High	500 kbps/call
Best Effort	0	Normal	Remaining

Network Requirements

Per Agent Bandwidth Requirements:
├─ Voice: 100 kbps (upload/download)
├─ Signaling: 10 kbps
├─ Desktop App: 50 kbps
├─ CRM Integration: 25 kbps
└─ Total: ~200 kbps per concurrent call

For 100 concurrent agents:
Total required bandwidth: 20 Mbps (with 20% overhead = 24 Mbps)

4.3 Jitter Buffer Configuration¶

Adaptive Jitter Buffer Settings:
├─ Minimum: 20ms
├─ Maximum: 200ms
├─ Target: 60ms
└─ Adaptation: Dynamic based on network conditions

4.4 Echo Cancellation¶

Built-in Echo Cancellation: - Webex platform provides automatic echo cancellation - ITU-T G.168 compliant - No configuration required

Troubleshooting Echo Issues: 1. Check agent headset quality 2. Verify acoustic environment 3. Review audio settings in desktop app 4. Test with alternate audio device

5. Dial Plan Design¶

5.1 Outbound Dialing¶

Dial Plan Rules

Pattern	Description	Action
9 + 1 + 10 digits	US/Canada long distance	Strip 9, route external
9 + 10 digits	US/Canada local	Strip 9, route external
9 + 011 + intl	International dialing	Strip 9, route external
4 digits	Internal extensions	Route to Webex Calling
911	Emergency services	Route direct, alert security
*xx	Feature codes	Platform features

Example Dial Plan Configuration:

RULE 1: Emergency
Pattern: 911
Action: Route immediately, no prefix, notify security

RULE 2: Internal Extensions
Pattern: [1-9]XXX (4 digits)
Action: Route to Webex Calling internal

RULE 3: Local Calls
Pattern: 9 + [2-9]XX-[2-9]XX-XXXX
Action: Strip leading 9, route to PSTN

RULE 4: Long Distance
Pattern: 9 + 1 + [2-9]XX-[2-9]XX-XXXX
Action: Strip leading 9, route to PSTN

RULE 5: International
Pattern: 9 + 011 + X+
Action: Strip leading 9, route to PSTN

5.2 Caller ID Management¶

Outbound Caller ID Strategy

Scenario	Caller ID Displayed	Configuration
Agent outbound call	Main company number	Default ANI
Callback from queue	Original customer number	Preserve ANI
Supervisor call	Supervisor direct number	Extension ANI
Emergency call	Site physical address	E911 ANI

Caller ID Format:

Format: +1-XXX-XXX-XXXX (E.164)
Example: +1-408-XX5-0100

Components:
+ = International prefix
1 = Country code (US/Canada)
408 = Area code
XX5-0100 = Local number

6. Emergency Services (E911)¶

6.1 E911 Configuration¶

Location Registration

Office Locations:
├─ HQ Campus (Building A)
│   ├─ Address: 123 Main St, San Jose, CA 95110
│   ├─ Emergency Contact: Security Desk
│   └─ Phone: +1-408-XX5-9111
│
├─ Regional Office (Building B)
│   ├─ Address: 456 Oak Ave, Austin, TX 78701
│   ├─ Emergency Contact: Facilities
│   └─ Phone: +1-512-XX5-9111
│
└─ Remote Agents
    ├─ Address: Agent's registered home address
    ├─ Emergency Contact: Agent's registered info
    └─ Phone: Agent's local 911 center

E911 Call Flow

Agent dials 911
    ↓
System identifies:
├─ Agent location (IP subnet or registered address)
├─ Closest PSAP (Public Safety Answering Point)
└─ Emergency callback number
    ↓
Routes to appropriate 911 center
    ↓
Simultaneously alerts:
├─ Corporate security desk
├─ Supervisor
└─ Emergency response team

6.2 Remote Agent E911¶

Requirements for Remote Workers: - Agents must register home address in system - System validates address against E911 database - Agents must update address if working from different location - Popup reminder every 90 days to confirm address

Testing Protocol:

Quarterly E911 Testing:
☐ Test call to non-emergency line (not 911!)
☐ Verify correct location information sent
☐ Confirm callback number accuracy
☐ Document test results
☐ Update any discrepancies

7. Call Recording Architecture¶

7.1 Recording Infrastructure¶

┌──────────────────┐
│   Active Call    │
│   (Agent+Caller) │
└────────┬─────────┘
         │
         │ RTP Stream
         │
┌────────▼─────────┐
│  Media Forking   │
│  (Real-time)     │
└────────┬─────────┘
         │
    ┌────┴────┐
    │         │
    ▼         ▼
┌────────┐ ┌────────────┐
│ Call   │ │ Recording  │
│ Audio  │ │ Metadata   │
│        │ │ (ANI, DNIS,│
│        │ │ Agent ID)  │
└───┬────┘ └─────┬──────┘
    │            │
    └─────┬──────┘
          │
    ┌─────▼──────┐
    │   Cloud    │
    │  Storage   │
    │ (Encrypted)│
    └────────────┘

7.2 Recording Policies¶

Recording Rules

Queue/Type	Recording Policy	Retention	PCI Compliance
Sales	100% of calls	90 days	No
Support	100% of calls	1 year	No
Billing	100% of calls	7 years	Yes - Pause on payment
Collections	100% of calls	7 years	Yes
Quality sampling	Random 10%	30 days	No

PCI-DSS Compliance:

When customer provides payment card:
1. Agent clicks "Pause Recording" button
2. Recording pauses immediately
3. Agent collects payment information
4. Transaction completes
5. Agent clicks "Resume Recording"
6. Recording resumes with notation in metadata

7.3 Storage and Retention¶

Storage Requirements Calculation:

Recording Storage per Agent per Day:
├─ Average call duration: 6 minutes
├─ Calls per day: 40 calls
├─ Total recorded time: 240 minutes = 4 hours
├─ Audio file size: ~3.6 MB per hour (G.711)
└─ Daily storage: ~14 MB per agent

For 500 agents over 1 year:
500 agents × 14 MB × 365 days = 2.5 TB

Data Lifecycle Management:

Timeline:
├─ 0-30 days: Hot storage (immediate access)
├─ 31-90 days: Warm storage (retrieval within minutes)
├─ 91 days-7 years: Cold storage (archive, retrieval within hours)
└─ 7+ years: Automated deletion (per retention policy)

8. Call Metrics and Monitoring¶

8.1 Key Telephony Metrics¶

Metric	Target	Alert Threshold
Call Setup Success Rate	>99%	<98%
Post-Dial Delay	<3 seconds	>5 seconds
Voice Quality (MOS)	>4.0	<3.5
Packet Loss	<1%	>2%
Jitter	<30ms	>50ms
Latency (Round-trip)	<150ms	>200ms

8.2 Real-Time Monitoring Dashboard¶

Key Indicators:

┌─────────────────────────────────────────────┐
│  Real-Time Telephony Dashboard              │
├─────────────────────────────────────────────┤
│  Active Calls: 247                          │
│  Trunk Utilization: 65% (195/300)           │
│  Average MOS Score: 4.2                     │
│  Failed Calls (last hour): 3 (0.5%)         │
│  Emergency Calls Active: 0                  │
└─────────────────────────────────────────────┘

8.3 Troubleshooting Tools¶

Built-in Diagnostics: - Real-time call quality monitoring - SIP ladder diagrams - RTP stream analysis - Network path visualization

Common Issues and Resolution:

Symptom	Likely Cause	Resolution
One-way audio	Firewall/NAT issue	Check firewall rules, verify RTP ports open
Choppy audio	Packet loss/jitter	Verify QoS, check network congestion
Echo	Poor acoustics	Check headset, verify echo cancellation
Call drops	Network instability	Review network stability, check bandwidth

9. Security Considerations¶

9.1 SIP Security¶

Security Measures: - TLS 1.2+ for SIP signaling encryption - SRTP for media encryption - SIP authentication via digest authentication - Rate limiting to prevent DoS attacks - Geo-blocking for international fraud prevention

9.2 Toll Fraud Prevention¶

Protection Strategies:

1. Outbound Call Restrictions:
   ├─ Block premium rate numbers (900, 976)
   ├─ Block international by default (whitelist only)
   ├─ Limit call duration (4-hour maximum)
   └─ Alert on unusual patterns

2. Authentication:
   ├─ Agent must authenticate before placing calls
   ├─ Supervisor approval for international calls
   └─ Two-factor authentication for admin changes

3. Monitoring:
   ├─ Real-time spend tracking
   ├─ Alert on spend threshold ($500/hour)
   └─ Automatic block on fraud detection

10. Telephony Testing Strategy¶

10.1 Pre-Production Testing¶

Test Scenarios:

☐ Inbound call routing to correct queues
☐ Outbound dialing (local, long distance, international)
☐ Call transfer (blind and attended)
☐ Conference calling
☐ Call recording (start, pause, resume, stop)
☐ Hold and resume
☐ DTMF tone recognition
☐ Voice quality across different codecs
☐ Failover scenarios
☐ Emergency (E911) routing
☐ After-hours routing
☐ Overflow routing
☐ CRM screen pop integration

10.2 Load Testing¶

Capacity Validation:

Test Profile:
├─ Concurrent calls: 500
├─ Duration: 4 hours
├─ Call arrival rate: Poisson distribution
├─ Average call duration: 5 minutes
└─ Expected behavior: No degradation

Success Criteria:
├─ All calls completed successfully
├─ Voice quality MOS > 4.0
├─ No dropped calls
├─ System response time < 3 seconds
└─ CPU/memory utilization < 80%

Validation Checklist¶

Before going live: