Agent Waits Full Second to Send Number? Explained

No, your agent does not wait a full second to send the number. In standard SIP (Session Initiation Protocol) call signaling, the delay between the agent pressing “send” and the number reaching the carrier is typically 10–300 milliseconds on a healthy network. The “full second” you observe is almost always a combination of network latency, carrier processing, or a misconfigured timer — not an intentional pause. This article explains what actually happens in that gap, why it can stretch to a second or more, and how to diagnose and fix it.

Background: How Call Setup Signaling Works

When an agent places a call through a softphone or desk phone, the device sends a SIP INVITE request to a call server (like Asterisk, FreeSWITCH, or a cloud PBX). The server routes it to a trunk provider, which forwards it to the destination carrier. Each hop adds latency. On a local network, that’s under 1 ms per hop; over the public internet, it’s typically 20–80 ms per round trip. A full second is far beyond normal, so the cause is usually not the agent’s device itself but something in the path.

What the “Send” Button Actually Does

When the agent clicks “Send” (or presses Enter) in a softphone, the client immediately constructs a SIP INVITE message. It does not intentionally delay. However, several factors can introduce a pause before the packet leaves the device:

• DNS resolution: If the softphone needs to resolve the PBX hostname and the DNS server is slow, that can add 100–500 ms.
• Authentication: If the PBX challenges the INVITE with a 401/407 response, the phone must re-send with credentials. Each challenge-response round trip adds latency.
• NAT keep-alive: Some clients send a keep-alive packet and wait for a response before sending the INVITE, which can add a full second if misconfigured.
• Registration expiry: If the phone’s registration has expired, it may attempt to re-register before placing the call, adding several seconds.

Common Causes of a One-Second Delay

A consistent ~1.0 second delay is often a specific timer, not random jitter. The most common culprits are:

• T1 timer (SIP INVITE retransmission): Default is 500 ms. If the first INVITE is lost, the client waits 500 ms before retransmitting. If the second is also lost, it waits 1 second. So a delay of exactly 1 second often means the first INVITE was dropped.
• DNS timeout: If the first DNS query times out (often 1–2 seconds), the client retries. This can produce a predictable 1-second delay.
• Proxy response delay: The PBX may be waiting for a response from an upstream carrier that is slow to answer. This is not the agent’s device.

How to Diagnose the Delay

To find where the second goes, capture the call flow. You can use Wireshark with SIP and RTP filters, or check the PBX logs. Look at the timestamps:

1. Note when the agent clicks “Send” (you can use a screen recording or a packet capture on the agent’s PC).
2. Check when the first SIP INVITE leaves the agent’s device.
3. Check when the PBX receives it.
4. Check when the PBX sends a 100 Trying or 180 Ringing back.
5. Compare timestamps. If the gap is between click and packet leaving the device, it’s a client or OS issue. If it’s between device and PBX, it’s network. If it’s between PBX and carrier, it’s the trunk.

Fixes and Optimizations

Depending on the cause, here are practical steps:

• DNS: Use a local DNS cache or IP address for the PBX. Ensure DNS servers respond in under 50 ms.
• Authentication: Some PBXs allow IP-based authentication to skip the challenge-response. If security permits, enable that.
• NAT keep-alive: Disable any “send keep-alive before call” option in the softphone.
• Network: Check for packet loss on the WAN link. Even 1% loss can trigger retransmission timers.
• PBX configuration: Reduce SIP T1 timer to 200 ms if you have a reliable LAN. But be careful: too low can cause false retransmissions.

Comparison: Softphone vs. Hardware Phone vs. WebRTC

Different client types have different default behaviors. Hardware phones often have a “pre-dial” delay because they need to wake the DSP. Softphones (like Zoiper, MicroSIP) are usually faster. WebRTC-based browsers (like Chrome) add a few hundred ms because they must negotiate a secure WebSocket connection to the PBX. If your agents use a web-based dialer, that’s a common source of a ~1 second delay.

Costs and Business Impact

While a 1-second delay might seem trivial, in high-volume call centers it adds up. If an agent makes 100 calls a day, that’s 100 seconds of dead time — about 1.7 minutes per agent per day. For a team of 50, that’s over an hour of lost productivity daily. More importantly, it can affect call quality metrics like Average Speed to Answer (ASA) and Service Level.

Frequently Asked Questions

Why does my softphone always wait exactly 1 second before dialing?

That pattern strongly suggests a SIP retransmission timer. The first INVITE is being lost or not acknowledged, so the client waits 500 ms, retransmits, waits another 500 ms, and gives up or sends again. Check for packet loss or firewall dropping SIP packets.

Is the delay on the agent’s side or the carrier’s side?

You can tell by looking at the PBX logs. If the PBX receives the INVITE immediately but the carrier response is slow, it’s the carrier. If the INVITE arrives late, it’s the agent’s network or device.

Can a VPN cause this delay?

Yes. VPNs add encryption overhead and may route traffic through distant servers. If agents use a VPN, try a split-tunnel configuration that sends only PBX traffic through the VPN, or test without VPN to compare.

Does using a headset vs. handset affect the delay?

No. The delay is in the signaling path, not the audio path. The headset only affects audio quality, not the time to send the number.

How do I measure the delay precisely?

Use Wireshark on the agent’s PC. Filter for SIP and look at the time between the user pressing the dial button (you can correlate with a screen recording) and the first INVITE packet. That gives you the client-side delay.

Is a 1-second delay acceptable for call centers?

It’s not ideal but often tolerable. However, if it’s consistent, it indicates a fixable issue. Most call centers aim for under 300 ms from click to ring.

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