SIP Traces

If you manage your own phone system (PBX) in your network, you will be asked to provide a SIP trace to SureVoIP Support to help troubleshoot call set up or call quality issues.

Asterisk and FreeSWITCH

As Asterisk and FreeSWITCH systems generally run on Linux it is very straight-forward to gather a SIP trace.

You will need:

  • root access
  • tcpdump to be installed

To gather a SIP trace with signalling only, run:

tcpdump -nnp -w siptrace.pcap -i any -s 0 port 5060

This captures only port 5060, which is the default SIP port. Change this if your system uses a different port. The file will be called siptrace.pcap and will be saved in your current directory.

Press Ctrl+C to stop the trace once you have made your test call and send the file to SureVoIP Support.

If you are troubleshooting voice quality you may be asked to provide a call trace including RTP.

You will need:

  • root access
  • tcpdump to be installed
  • you will need to know the RTP ports used by your system
    • Asterisk default is ports 10000-20000
    • FreeSWITCH default is ports 10000-40000

To capture SIP and RTP for Asterisk using the default ports run:

tcpdump -nnp -w test.pcap -i any -s 0 port 5060 or portrange 10000-20000

or to capture SIP and RTP for FreeSWITCH using the default ports run:

tcpdump -nnp -w test.pcap -i any -s 0 port 5060 or portrange 10000-40000

Make your test call and when finished press Ctrl+C to end the capture and then send the file to SureVoIP Support.

Other Phone Systems

All reputable phone systems will have a method to acquire SIP traces for troubleshooting purposes. Please consult your vendor's documentation or consult with a qualified professional to assist.

For PBX software running on Microsoft Windows, we suggest using Wireshark to obtain SIP traces.

There may be times when a SIP trace from a handset is required.

Other Useful Information

There are several tools you can use to obtain SIP traces, such as: