Time to First Media
Understand Time to First Media — how CallMeter measures the delay from call answer to the first RTP packet, and what slow media setup reveals about NAT, codec negotiation, and network configuration.
Time to First Media measures the delay between the SIP 200 OK (call answered) and the arrival of the first RTP media packet, in milliseconds. The call is "connected" at 200 OK, but audio does not begin until RTP packets start flowing. This gap represents the time needed for media negotiation, NAT traversal, and the first packet to transit the network.
Think of it as the pause between a phone call being answered and actually hearing the other person speak. The call is connected, but there is a brief moment of silence before audio begins. In a well-configured system, this pause is barely perceptible.
One-shot metric
Time to First Media is captured once per call. It appears in the Call Timing Overview cards on the endpoint detail page.
How It Works
After the SIP 200 OK is exchanged, both endpoints know each other's media parameters (codecs, IP addresses, ports) from the SDP negotiation. They then begin sending RTP packets. CallMeter timestamps the first RTP packet received after the 200 OK:
Time to First Media = first_rtp_at - answer_at
This interval includes:
- SDP processing and codec initialization
- ICE connectivity checks (if ICE is used)
- DTLS handshake for SRTP key exchange (if encrypted)
- NAT/firewall hole punching
- First RTP packet network transit time
Why It Matters
Time to First Media is a critical diagnostic metric that reveals problems invisible to SIP signaling metrics:
- NAT traversal issues — Long time-to-first-media often indicates NAT binding or hole-punching delays
- SRTP negotiation problems — DTLS handshake failures or slow key exchange delay media startup
- Codec initialization — Some codecs require setup time before producing the first frame
- Firewall rules — RTP may be blocked while SIP succeeds, creating a connected call with no audio
For enterprise testing, this metric catches the dreaded "one-way audio" and "no audio" scenarios that SIP success alone does not reveal.
Common Causes of Slow Time to First Media
| Cause | Explanation |
|---|---|
| NAT traversal delay | ICE connectivity checks or STUN binding requests taking time |
| DTLS-SRTP handshake | Encryption key exchange adding latency before media can flow |
| Firewall blocking RTP | SIP allowed through but RTP/UDP packets blocked or delayed |
| Codec initialization | Complex codecs requiring setup before first frame |
| Symmetric NAT | Both endpoints behind symmetric NAT, requiring TURN relay setup |
| SBC media processing | Session Border Controller setting up media relay path |
How to Fix It
- Check for one-way or no audio — If time to first media is very high or infinite, the RTP path may be completely blocked. Verify that firewalls allow UDP on the RTP port range.
- Review ICE configuration — If ICE is used, check whether STUN/TURN servers are responsive and reachable. Slow or failed ICE candidates delay media.
- Test SRTP negotiation — If using encrypted media, verify that the DTLS handshake completes quickly. Certificate exchange delays can add hundreds of milliseconds.
- Inspect NAT behavior — Symmetric NAT is the most problematic for RTP. If endpoints are behind symmetric NAT, ensure TURN relay is available.
- Monitor SBC media setup — If an SBC sits in the media path, its media relay setup time contributes directly to this metric.
Related Metrics
- Call Setup Time — Signaling duration before media setup begins
- Post-Dial Delay (PDD) — Pre-answer delay; time to first media adds to the total caller wait
- Round Trip Time — Network latency that contributes to media setup delay
- Call Result — A successful SIP setup with no media indicates an audio path problem
RFC Reference
RTP media transmission is defined in RFC 3550 (RTP: A Transport Protocol for Real-Time Applications). The relationship between SIP signaling completion and RTP startup is described in RFC 3264 (An Offer/Answer Model with the Session Description Protocol). ICE connectivity checks that affect time-to-first-media are defined in RFC 5245 (Interactive Connectivity Establishment).
Call Setup Time
Understand Call Setup Time — how CallMeter measures the total SIP signaling duration from INVITE to 200 OK, thresholds, and what affects the complete call establishment.
Call Duration
Understand Call Duration measurement — how CallMeter tracks the connected time of a test call, and what duration reveals about test scenario execution and SLA compliance.