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.
As Asterisk and FreeSWITCH systems generally run on Linux it is very straight-forward to gather a SIP trace.
You will need:
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:
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.
Please see 3CX SIP Trace guide (external page).
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.
There are several tools you can use to obtain SIP traces, such as: