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 ------------------------------------------------------------------------- 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
