John Lange said: > I have no problem using the sample SIPP uac_pcap on the sending side, > but on the receiving end, how do I build a SIPP Scenario that will > answer the call and capture the audio stream? I see the play_pcap_audio, > but no record_pcap_audio. What is the recommended way to achieve this? >
[EMAIL PROTECTED] said: > I believe you will have to capture packets with a tool like tcpdump or > ethereal/wireshark or maybe rtptools > (http://www.cs.columbia.edu/IRT/software/rtptools/). With some well > defined capture filters, you'll get what you want. I am working on the same problem, and I modified uac_pcap.xml with a listening part with tcpdump, a small problem is that Sipp don't seem to have a variable for the local receiving port. See enclosed script-bits. > I would like to compare the captured audio to the original to determine > loss etc. with tcpdump you have a pcap-format that you can compare bit for bit with the origingal audio, but that is hardly productive. rtptools also seems to keep this on a packet level, so we need to find a tool to transform pcap to audio or a tool to store audio directly, and then to find a nice simple scriptable tool to compare the audio quality. cheers Olav my start of a solution <!-- Play a pre-recorded PCAP file (RTP stream) --> <nop> <action> <exec play_pcap_audio="pcap/g711a.pcap"/> <exec command="tcpdump-vent 9 -s 0 -w uac_echo_[remote_ip]_6000_`date +%F_%T`.pcap src host [remote_ip] and dst port 6000"/> </action> </nop> The tcpdump-vent script is like this #!/bin/sh vent=$1 shift tcpdump $* & sleep $vent kill %1 exit 0 > On 12/5/06, John Lange <[EMAIL PROTECTED]> wrote: > > I would like to use SIPP to test an end-to-end VoIP environment for > > audio call quality under load. > > > > This schenario would involve two Asterisk boxes connected via the PSTN > > like follow: > > > > SIPP send <--> Asterisk <--> T1 PSTN <--> Asterisk <--> SIPP receive > > > > I have no problem using the sample SIPP uac_pcap on the sending side, > > but on the receiving end, how do I build a SIPP Scenario that will > > answer the call and capture the audio stream? I see the play_pcap_audio, > > but no record_pcap_audio. What is the recommended way to achieve this? > > > > I would like to compare the captured audio to the original to determine > > loss etc. > > > > John > > > > Hi John, > > I believe you will have to capture packets with a tool like tcpdump or > ethereal/wireshark or maybe rtptools > (http://www.cs.columbia.edu/IRT/software/rtptools/). With some well > defined capture filters, you'll get what you want. > > Then you will have to do some further processing to get what you need, > which is not so clear from your post. > > Good luck! > > Juan > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > _______________________________________________ > Sipp-users mailing list > [email protected] > https://lists.sourceforge.net/lists/listinfo/sipp-users > ------------------------------------------------------------------------- Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT & business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV _______________________________________________ Sipp-users mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/sipp-users
