3.3 Audio/Video Codecs & DTMF Relay

3.3 Audio/Video Codecs & DTMF Relay

Enterprise collaboration networks rely on coder-decoders (codecs) to convert analog voice and video signals into digital packetized streams for transport across IP networks. Simultaneously, networks must reliably convey user digit input—Dual-Tone Multi-Frequency (DTMF) signaling—to support Interactive Voice Response (IVR) systems, automated attendants, and voicemail navigation.

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1. Enterprise Audio Codecs

Audio codecs differ in their compression algorithms, bandwidth utilization, processing latency, and perceptual audio fidelity. The standard metric for voice quality is the Mean Opinion Score (MOS), ranging from 1.0 (unacceptable) to 5.0 (crystal clear / toll quality = ~4.0).

Detailed Codec Breakdown

G.711 (u-law and A-law)

• Algorithm: Pulse Code Modulation (PCM) without compression.
• Bitrate: 64 kbps.
• Sampling Rate: 8 kHz with 8-bit sample resolution (Nyquist theorem: 2 × 4 kHz acoustic spectrum = 8,000 samples/sec × 8 bits = 64,000 bps).
• Flavors:
  • u-law (PCMU): Standard in North America and Japan (logarithmic companding curve).
  • A-law (PCMA): Standard in Europe and the rest of the world.
• Packetization: Default 20 ms frame size = 160 bytes audio payload per packet.
• Bandwidth Calculation: $\text{Voice Payload} = \frac{64{,}000\text{ bps} \times 0.020\text{ s}}{8\text{ bits/byte}} = 160\text{ bytes}$ $\text{Total Packet} = 160\text{ bytes (payload)} + 40\text{ bytes (IP/UDP/RTP)} = 200\text{ bytes}$ $\text{Layer 3 Bandwidth} = \frac{200\text{ bytes} \times 8\text{ bits/byte}}{0.020\text{ s}} = 80.0\text{ kbps}$ $\text{Ethernet Bandwidth} = \frac{(200 + 14\text{ Ethernet} + 4\text{ FCS}) \times 8}{0.020\text{ s}} = 87.2\text{ kbps}$
• MOS Score: 4.1.

G.729 (G.729a and G.729b)

• Algorithm: Conjugate-Structure Algebraic-Code-Excited Linear Prediction (CS-ACELP).
• Bitrate: 8 kbps.
• Sampling Rate: 8 kHz; 10 ms speech frames, typically packetized as two frames per packet (20 ms payload = 20 bytes).
• Variants:
  • G.729a: Computationally simplified Annex A version, reducing DSP processing load by ~50% with negligible audio degradation.
  • G.729b: Annex B adds Voice Activity Detection (VAD) and Comfort Noise Generation (CNG) to suppress transmission during pauses, saving ~30–40% bandwidth.
• Layer 3 Bandwidth (20 ms): $(20\text{ bytes} + 40\text{ bytes header}) \times 8 / 0.020\text{ s} = 24.0\text{ kbps}$.
• MOS Score: 3.9.

G.722 (Wideband / HD Voice)

• Algorithm: Sub-band Adaptive Differential Pulse Code Modulation (SB-ADPCM).
• Bitrate: 64 kbps (with operating modes at 56 kbps and 48 kbps).
• Sampling Rate: 16 kHz (captures frequencies from 50 Hz to 7,000 Hz, compared to 300 Hz–3,400 Hz for G.711).
• Bandwidth: Identical to G.711 (80 kbps at Layer 3 with 20 ms ptime), but delivers substantially richer "High-Definition Voice" acoustics.
• MOS Score: 4.2.

Opus (RFC 6716)

• Architecture: Highly versatile open standard combining Skype's SILK (voice compression) and CELT (ultra-low latency music/audio).
• Bitrate: Adaptive from 6 kbps to 510 kbps.
• Sampling Rates: Fullband up to 48 kHz (supports narrowband 8 kHz, mediumband 12 kHz, wideband 16 kHz, super-wideband 24 kHz, and fullband 48 kHz).
• Packetization: Dynamically variable frame sizes from 2.5 ms up to 60 ms.
• Use Case: Native codec for Webex Calling, Webex App, WebRTC, and modern Cisco IP Phone 8800 series firmware.

iLBC (Internet Low Bitrate Codec, RFC 3951)

• Bitrate & Frame Size: 13.33 kbps (30 ms frame = 400 bits) or 15.2 kbps (20 ms frame = 304 bits).
• Resilience: Employs independent frame encoding. A lost packet does not corrupt adjacent frames, making it exceptionally stable across lossy public Internet or satellite connections.
• MOS Score: 3.8.

Audio Codec Reference Table

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2. Enterprise Video Codecs

H.264 AVC (Advanced Video Coding, MPEG-4 Part 10)

The ubiquitous standard across Cisco Room OS endpoints, Jabber, and Webex devices:

• Profiles:
  • Constrained Baseline Profile (CBP): Uses I-slices and P-slices with CAVLC entropy coding. Low computational complexity and low latency; ideal for early video endpoints.
  • Main Profile: Introduces B-slices (bidirectional predictive coding) and CABAC (Context-Adaptive Binary Arithmetic Coding).
  • High Profile: Introduces the 8×8 transform and intra-prediction and custom quantization scaling matrices, giving noticeably better compression than Baseline at the same quality, at the cost of more encoder and decoder processing.

H.265 HEVC (High Efficiency Video Coding)

• Efficiency: Delivers approximately 50% better compression than H.264 High Profile.
• Architecture: Replaces macroblocks with flexible Coding Tree Units (CTUs) up to 64×64 pixels.
• Application: Essential for high-resolution 4K/60fps video collaboration, Cisco Webex Room Panorama systems, and bandwidth-constrained WAN links.

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3. DTMF Relay Mechanisms

DTMF signaling generates dual frequencies per digit (e.g., dialing '5' generates 770 Hz and 1336 Hz). How these digits traverse the network is critical:

In-Band DTMF

• DTMF tones are sampled directly into the voice stream as standard uncompressed audio (audio/pcmu).
• Failure Mode with Compression: When passed through lossy perceptual codecs like G.729 or iLBC, speech compression algorithms treat the pure dual tones as background noise or alter their frequency harmonics. The remote IVR fails to recognize dialed digits. In-band DTMF is only reliable over uncompressed G.711.

Out-of-Band DTMF Methods

1. RFC 2833 / RFC 4733 (Named Telephony Events - RTP-NTE)

• Standard: IETF standard dynamic RTP payload (telephony-event, commonly payload type 101).
• Operation: When a key is pressed, the endpoint stops sending audio and transmits discrete RFC 2833 RTP packets containing:
  • Event ID: 0–9, *, #, A–D (e.g., digit '5' = event 5).
  • Volume: 6-bit signal level in dBm0.
  • Duration: 16-bit integer representing timestamp duration.
  • End of Event (E) bit: Set to 1 on the final 3 redundant packets to ensure delivery even under packet loss.

2. Keypad Markup Language (KPML, RFC 4730)

• Standard: XML-based signaling carried within SIP.
• Operation: The call agent (CUCM) sends a SUBSCRIBE request for the kpml event package. When the user presses a digit, the phone sends a NOTIFY request containing an XML body:

  <?xml version="1.0" encoding="UTF-8"?>
  <kpml-response version="1.0" code="200" text="OK" digits="5" tag="dtmf"/>
  

• Supported natively by Cisco Unified IP Phones registered to CUCM.

3. SIP INFO

• Carries DTMF in the body of a mid-call INFO message (application/dtmf-relay or application/dtmf).
• Hop-by-hop signaling. Not widely supported by public carrier SIP trunks.

4. Cisco Unsolicited NOTIFY

• Gateway sends a NOTIFY message containing the DTMF digit without waiting for a prior SUBSCRIBE.

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4. DTMF Interworking on Cisco CUBE & CUCM

In real-world collaboration environments, endpoints frequently use mismatched DTMF mechanisms. For example, internal Cisco IP Phones use KPML to communicate with CUCM, while the external Session Border Controller (CUBE) connects to an Internet Telephony Service Provider (ITSP) that mandates RFC 2833 (RTP-NTE).

Asymmetric DTMF on Cisco IOS XE CUBE

Cisco CUBE supports asymmetric DTMF negotiation, receiving KPML on the LAN call leg and converting it to RFC 2833 on the WAN call leg (or vice versa) without needing DSP transcoding resources.

! Inbound dial-peer from CUCM
dial-peer voice 101 voip
 description Inbound Leg from CUCM
 session protocol sipv2
 incoming called-number .
 codec g711ulaw
 dtmf-relay sip-kpml rtp-nte
!
! Outbound dial-peer to ITSP
dial-peer voice 201 voip
 description Outbound Leg to Carrier SIP Trunk
 destination-pattern +1[2-9]..[2-9]......
 session protocol sipv2
 session target ipv4:198.51.100.1
 codec g711ulaw
 dtmf-relay rtp-nte

MTP Allocation for DTMF Conversion in CUCM

If an endpoint or trunk cannot perform native asymmetric DTMF negotiation, CUCM must insert a Media Termination Point (MTP):

• When a SIP trunk only supports RFC 2833, but an endpoint only supports KPML or in-band audio, CUCM dynamically allocates an MTP from the Media Resource Group (MRG).
• The MTP terminates the RTP stream, extracts the RFC 2833 telephony-event packets, and generates KPML SIP NOTIFY messages toward CUCM (or vice versa).
• If no MTP is available in the endpoint's Media Resource Group List (MRGL), DTMF digits fail to register with remote IVR systems.