Not sure what exactly you are asking.

You can use play_pcap_audio to replay a single RTP stream. The .pcap file to be played must not contain anything else but that stream. SIPp then changes the source and destination addresses and ports according to actual values used locally (from auto-media-port in the scenario and from the mi command line parameter) but keeps the rest (ssrc, sequence numbers) unchanged.


So to replicate a failed call scenario based on a pcap, you have to

- use a display filter which shows only that single RTP stream and use "export selected packets -> displayed" to save them into a new .pcap file which you would then use for play_pcap_audio,

- use a display filter like "sip.Call-ID==your_call-id" to show only messages belonging to the SIP dialog in question and use "export selected packets -> displayed" to save them into another new .pcap(ng) file

- disable the SIP dissector, set Wireshark to display non-dissected data as text and run tshark from command line:

tshark.exe -r your_single_SIP_dialog_file_name.pcapng -T fields -e data.text > your_file_name.txt

to extract the SIP conversation from the pcap(ng) file into a text form (switching on "display SIP text" in SIP preferences doesn't currently work well with tshark)

- edit your_file_name.txt to add the XML tags around the messages to convert it into a SIPp scenario file

Pavel


Dne 11.4.2018 v 13:59 Franklin Angulo napsal(a):

Hi. I am using SIPp to create test scenarios that simulate real SIP communication scenarios between server and client that have generated errors when installing a real VoIP system. I have a capture .pcapng of Wireshark and I want that based on this capture I can generate a SIP scene with which I can perform my tests in the laboratory with my SIP phone or the server depending on the case that occurs. What I do first is convert the .pcapng to .pcap format to include the .pcap in the statement <exec play_pcap_audio = "pcap / File name.pcap" /> . My scenario is:

IP Phone IP Server

IP Phone Invite ---> IP Server
IP Phone <- 100 IP Server
IP Phone <- 407 IP Server
IP Phone ACK ---> IP Server
IP Phone Invite ---> IP Server
IP Phone <- 100 IP Server
IP Phone <- 180 IP Server
IP Phone <------------------------------------ RTP (G711U)
IP Phone <- 200 OK SDP G711U IP Server
IP Phone <- 180 IP Server
IP Phone Cancel ---> IP Server
IP Phone <- 481 Call IP Server
IP Phone ACK ---> IP Server
IP Phone <- 200 OK SDP G711U IP Server
IP Phone <- By IP Server

Is it possible to generate this scenario with the audio and the SIP / SDP that the .pcapng file from Wireshark gives me, having compiled the SIPp with ./configure --with-pcap or does it have to add another module?

Thank you very much for the information!

Carefully greetings,

Franklin



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