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
