Adjacent Node
Networking, explained. No BS.

VoIP Basics

What It Is

Voice over IP carries call signaling and media over packet networks. Signaling sets up, changes, and tears down calls. Media usually flows as RTP or SRTP between phones, gateways, SBCs, conferencing services, or cloud calling platforms.

Modern voice work is less about one PBX and more about end-to-end readiness: switching, PoE, VLANs, DHCP options, DNS, QoS, firewall policy, NAT traversal, SBCs, certificate trust, and cloud service reachability.

Core Pieces

Piece Role
SIP Common call signaling protocol
RTP Real-time media transport
SRTP Encrypted RTP media
RTCP Media quality reporting and control
SBC Session border controller for policy, NAT, security, and interconnect
Codec Converts audio to packet payload
Jitter buffer Smooths packet delay variation
PoE Powers phones and access points

Modern note: Phone registration working does not prove media works. Signaling and RTP/SRTP often take different paths and use different ports.

Common Codecs

Codec Typical Bitrate Notes
G.711 64 kbps High quality, common LAN and PSTN interop
G.722 64 kbps Wideband audio
G.729 8 kbps Low bandwidth, licensing and quality tradeoffs
Opus Variable Common in modern collaboration apps
iLBC 13.33 or 15.2 kbps Legacy packet-loss resilient codec

Watch out: Codec bitrate is not total wire bandwidth. Add IP, UDP, RTP, Ethernet, VLAN, tunneling, and security overhead.

Call Setup Path

Step What Happens
Power Phone receives PoE or local power
VLAN Phone learns voice VLAN via CDP, LLDP-MED, or manual config
Addressing Phone gets IP, gateway, DNS, and vendor options
Provisioning Phone downloads config or contacts cloud service
Registration Phone registers to call control
Signaling Call is set up with SIP or another signaling protocol
Media RTP or SRTP flows between endpoints or media relays

Access Switch Port

Cisco-style phone plus PC port:

interface GigabitEthernet1/0/10
 description Desk phone plus workstation
 switchport mode access
 switchport access vlan 20
 switchport voice vlan 30
 spanning-tree portfast
 power inline auto
 auto qos voip cisco-phone

Manual QoS trust pattern:

interface GigabitEthernet1/0/10
 mls qos trust device cisco-phone
 mls qos trust cos

Notes:

  • Exact QoS commands vary by switch family and software release.
  • Trust the phone, not an arbitrary PC.
  • LLDP-MED is often better than vendor-only discovery in mixed environments.

PoE

Standard Common Name Notes
IEEE 802.3af PoE Older phones and low-power endpoints
IEEE 802.3at PoE+ Higher power phones, APs, cameras
IEEE 802.3bt PoE++ High-power APs, cameras, displays
Cisco pre-standard Inline power Legacy Cisco environments

Watch out: A switch can have enough per-port power but not enough total power budget for all ports.

QoS Targets

Traffic Common Marking Notes
Voice RTP DSCP EF 46 Low delay and jitter
Voice signaling CS3 or AF31 Keep consistent with local standard
Video media AF41 or service-specific Validate app guidance
Default data BE 0 Normal traffic
Scavenger CS1 Backups, guest, low priority

Design note: QoS must be applied where congestion happens: WAN edge, Wi-Fi, VPN headend, SD-WAN overlay, or provider handoff.

Cloud Calling And Meetings

Area What To Verify
DNS Service discovery and certificate names resolve correctly
Firewall Required ports and destinations are allowed
NAT Media paths, symmetric NAT behavior, and SBC policy
TLS Certificates and inspection bypass where required
QoS DSCP preservation or remarking through edge and WLAN
Wi-Fi Voice SSID, roaming, airtime, retries, and WMM
Monitoring MOS, packet loss, jitter, latency, and call detail records

Watch out: TLS inspection, proxying, or blocked UDP can force media relay paths and degrade meetings even when calls still connect.

Troubleshooting

Symptom Check Likely Cause
Phone will not boot PoE class, power budget, cabling No power or bad pair
No registration VLAN, DHCP, DNS, TFTP/cloud reachability Provisioning path broken
One-way audio NAT, ACL, RTP path, codec, SBC Signaling works, media path fails
Poor audio Loss, jitter, queue drops, Wi-Fi retries Congestion or RF issue
Calls drop SIP timers, firewall timeout, ALG, SBC logs Stateful middlebox issue
Video poor but voice fine Bandwidth, QoS class, app policy Media class or capacity issue
Only cloud meetings fail URL/IP allowlists, UDP blocked, TLS inspection Service edge policy

Commands

show power inline
show cdp neighbors detail
show lldp neighbors detail
show interfaces switchport
show interfaces status
show policy-map interface
show access-lists
show logging | include PHONE|POWER|LINEPROTO

Expected clues:

  • Phone is in the voice VLAN and PC is in the data VLAN.
  • DHCP options and DNS point to the intended call services.
  • RTP/SRTP packets are not blocked by ACLs or NAT.
  • QoS class counters increment during calls.
  • Wi-Fi voice clients have acceptable RSSI, SNR, retry rate, and roaming behavior.

Watch Out

  • Do not trust call signaling as proof of media quality.
  • Do not let SIP ALG modify traffic unless you know it is required.
  • Do not ignore PoE budget during phone or AP refreshes.
  • Do not classify voice only at the LAN edge and forget the WAN or Wi-Fi bottleneck.
  • Do not assume cloud collaboration apps use the same ports and paths forever.

References