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LAUNCHTelephony

SIP Call Ladder

See the SIP signaling and audio quality of any call, and download its packet capture.

Telephony tab of a call in Logs, showing the SIP ladder

The SIP call ladder shows every SIP message exchanged on a call, in order, with timestamps. Use it to find where a failed call stopped, check the call’s audio quality, follow each warm transfer attempt, and download the packet capture.

To open it, select a call in Logs, then select the Telephony tab.

Ladder diagram

  • Vertical lines are the network elements on the call: your agent, Synthflow’s network hops, and the SIP carrier.
  • Horizontal arrows are SIP messages between them, such as INVITE, 100 Trying, 180 Ringing, 200 OK, and BYE.
  • Timestamps show when each message was sent, so you can see where time was spent.

Common patterns

SymptomPattern on the ladderWhat to check
Call does not connectINVITE with no responseThe agent’s number, and the SIP Domain if it is a custom number.
Call dropsAn unexpected BYENetwork timeouts, or issues at the carrier.
Authentication fails401 or 403 responseThe SIP credentials and the calling permissions on the account.
One-way audio200 OK, but RTP problemsCodec compatibility, the source and destination IPs, and your firewall rules.

For the meaning of a specific error code, see Telephony Errors.

Warm transfers

SIP Ladder with Warm transfer 1 selected, showing the INVITE from Synthflow to Twilio and the 100 response

When a call has a warm transfer, the ladder shows each transfer attempt separately. Cold transfers do not appear. Select the attempt from the dropdown next to SIP Ladder to see where a failed transfer stopped.

Pattern on the transfer attemptWhat it means
INVITE with no final responseThe destination never answered. Check the number and the transfer Timeout.
486 Busy HereThe destination line was busy.
404 Not Found or 484 Address IncompleteThe carrier could not route the number. Check it is in E.164 format.
403 ForbiddenThe carrier refused the call, often because of the caller ID you present.
200 OK, then an early BYEThe destination answered and hung up, or human detection ended the attempt.

Match the pattern against the transfer’s failure reason in the call’s Actions tab.

Call quality

The Telephony tab also reports the call’s audio quality, measured from RTCP. Check it first when a caller reports choppy or delayed audio, before you download a PCAP.

MetricWhat it tells you
MOSA score from 1.0 to 5.0 estimating how the call sounded overall. Target above 4.2.
Round-trip timeHow long audio takes to reach the far end and come back, in milliseconds. Target under 100 ms for regional traffic.
JitterHow much packet arrival times vary, in milliseconds. High jitter makes speech sound robotic or choppy.
Packet lossThe percentage of audio packets that never arrived.

A dash means the value is unknown, not zero. It appears when the remote endpoint sends no RTCP, which is common on the carrier leg, or when the audio did not pass through our media relay.

The same measurements are available through the API as telephony_rtp_* fields on Get a call and List calls, reported separately for the WAN (carrier) and LAN (internal) legs. For targets and how to measure on your own network, see Call quality.

PCAP download

Download PCAP button in the Telephony tab

The PCAP file is the full packet capture of the call’s signaling and audio. Select Download PCAP in the Telephony tab, then open the file in a packet analyzer such as Wireshark.

FAQ

Send the PCAP file, a screenshot of the SIP ladder, and the URL of the agent.