I looked at Playing audio file using sipp prior to posting this... I followed the instructions posted there. However, looking at a packet capture I see that typically my calls only contain 8-12 packets - not nearly enough to transfer 30s of audio that I created in my .WAV file. Additionally, wireshark doesn't detect anything other than SIP traffic - no RTP payloads.
Only the server side I see "Aborting call on UDP retransmission timout for Call-ID 'xxx..." I don't see any errors on the client side logs.
Are there any updated examples of how to push an audio file to sipp acting in uas mode from sipp acting in uac mode? I've played around with it for a couple of days and read the documentation at http://ift.tt/2c8gEcz but I haven't had much luck.
Can someone point me to a howto or example scenario file for both client and server? I would love to see something using rtp_echo to send played audio back to the client in order to simulate full dialogue.
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