you might want to make that -m 1 for now so that you can verify it works. Stat_delimiter: i change the default to ,Īlso, note that I have this for 2000 calls, at the default rate (10CPS). Trace_rtt and trace_err: I output the log files and performance report Mi: mi specifies your IP for sending media. Sudo: Why sudo? If you don't sudo this, it can't create the socket needed to send the audio/media. So I'm using my own scenario file, but the above params should work fine for you. Sudo sipp -s -sf /home/sipp-3.3/uac_pcap_g711.xml -m 2000 -mi -d 1200 -trace_rtt -trace_err -stat_delimiter ,
![pcap to wav converter pcap to wav converter](https://sc01.alicdn.com/kf/HTB1XbamDKySBuNjy1zdq6xPxFXa8/230978660/HTB1XbamDKySBuNjy1zdq6xPxFXa8.jpg)
if you can make that connection fine, the rest isn't so bad. So first, make sure you can correctly get your call to a softphone/phone. I banged my head on a wall for some time with this as well.
#Pcap to wav converter full#
More details on the full shell script I'm using: but that's outside the scope of your question. I did some more stuff in my automation framework (like transcribe the audio to text and compare against a known string). Now the script is setting those RTP.ssrc's to a output file, i'm calling sip_1call.rawįor my purpose I also wanted to convert that raw file to a wav, so I used sox: sox -t raw -r 8000 -v 4 -c 1 -U /jenkins/userContent/sip_1call.raw /jenkins/userContent/sip_1call.wav sudo tshark -n -r /jenkins/userContent/sip_1call.pcap -R rtp -R "rtp.ssrc = $ssrc" -T fields -e rtp.payload | tee payloadsĪt this point of my shell script, I'm running tshark again on the recorded pcap and taking that ssrc value and saying "find all of them as 'payload'" for payload in `cat payloads` do IFS=: for byte in $payload do printf "\\x$byte" > /jenkins/userContent/sip_1call.raw done done You would need to capture all the RTP.ssrc's and output them to a file and that can easily become raw audio again.
![pcap to wav converter pcap to wav converter](http://img.ibxk.com.br/77988/180184.jpg)
If you have one stream, you'd have one RTP ssrc value. What the ssrc is is, is an identifier of a RTP stream. I'm setting the variable ssrc to this action of using tshark to pull out the rtp ssrc value. The above records a packet capture for 20 seconds (the duration of the call going on at the same time) and outputs the packets as sip_1call.pcap ssrc=$(tshark -n -r /jenkins/userContent/sip_1call.pcap -R rtp -T fields -e rtp.ssrc -Eseparator=, | sort -u | awk 'FNR =1 ')
![pcap to wav converter pcap to wav converter](https://i.imgur.com/kH0zpte.png)
So really I have a shell script that does this: tshark -a duration:20 -w /jenkins/userContent/sip_1call.pcap I do this all via the command line so it can be automated. To do this, I stripped the RTP out of the pcap, then converted the rtp file to raw audio, and then to a wav. I was making a test call, and wanted to convert the pcap recorded to audio. I found several tutorials online and used them to build a test harness that automatically rebuilds the audio/rtp on a pcap, then makes a wav and transcribes the audio on that wav to text. However, if you want to automate this, and avoid the UI. click a stream it lists, and then 'analyize.' You could use Wireshark's UI to easily take the RTP from the pcap via the Menu: Telephony/RTP/ then show all streams. Sox is a audio analysis tool that is run from the command line.Īfter looking at some examples online of different tools, I pieced this together from other people's examples, with a few modifications.If I understand correctly, you have the pcaps, but you want to get the RTP from them?
#Pcap to wav converter install#
It's installed on centos boxes using yum install wireshark-gnome.
![pcap to wav converter pcap to wav converter](https://imag.malavida.com/mvimgbig/download-fs/free-video-to-mp3-converter-6902-2.jpg)
Tshark is a command line version of wireshark. I've been wanting to do a packet capture during the test and convert it back to audio afterwards, then do a wav comparison on the expected audio vs. But how do I know I get the right end point? In the past, I'd have the phone number I dial, record voice mail and send me an email, and the sipcli client would send over text to speech audio.īut I dont always have the luxury of being able to configure the phone number to voice mail. I currently drive automated SIP calls via SIPCLI and ruby for a variety of tests at work. After following a lot of different tutorials (some of which worked some of which didn't), I came up with a shell script using a couple tools to scrape a packet capture file, pull out the rdp packets, and then convert them back into audio.įor me this will be useful in automated testing.