FWIW, I had the exact same problem (Ok, it was an IVR that couldn't hear
the audio, but essentially the same thing).
After giving up trying to get it to work, I did wonder if my scenario XML
was negotiating a different audio codec from that being replayed in the
pcap file, but never went back to check. Worth looking into, though.
Greg
On Friday, 8 February 2013, Brian Warner wrote:
> Hi guys,
>
> I was given a task at my job to automate some SIP functionality. I'm
> using SIPP to do that. I can drive SIPP tests via Cucumber to our
> FreeSwitch box... i.e. I can get my desk phone to ring and verify that I'm
> getting the proper behavior to pass these functional tests.
>
> Now I want to verify that the other end picks up, and hears audio.
>
> I've tried dozens of guides on this online, setting up a pcap file, etc.
> I've tried the built in uac scenario, I've tried outputing that and
> modifying it with other people's ideas... the end result is this:
>
> If I drive a SIPP uac_pcap test to a softphone, and it rings, and I
> answer... I never hear the audio.
>
> My expectation, is that when I click "answer" on the softphone, I should
> hear that pcap audio file, wether it's a dtmf, or a voice msg. But i just
> get dead air.
>
> I'd first like to verify that this is actually sending audio, before go
> further.
>
> Currently I'm doing this:
>
> 1. On my local box, I call our Test Freeswitch in an integration env:
> ./sipp -s 1213 -sf uac_mine.xml -d 5000 -m 1 [FreeswitchIP]
>
> 2. The service there is a Virtual Number/account. It's set up to redirect
> the call to a different number - which I have a softphone configured to it.
>
> 3. The xml has been modified that after an ACK it does:
>
> <nop>
> <action>
> <exec play_pcap_audio="pcap/test_reverb.pcap"/>
> </action>
> </nop>
>
> When I run that command, sure enough it redirects the call to my
> softphone... if I manually answer I get dead air - when I'm expecting to
> hear the rtp audio from that pcap.
>
> Debugging:
> I've tried a few things:
> 1. I've tried using Phone 1, to call that virtual number on the FreeSwitch
> It rings Softphone 2, I answer.... Audio from my voice transmits just fine.
>
> So the flow between the call, freeswitch and the receiver is working.
>
> 2. I've also tried using the built in uac_pcap, considering perhaps my
> coded xml is bad. Yet the built in scenario also doesn't send any audio
> through... i just get silence.
>
> 3. Instead of hitting the integration FreeSwitch we have here, I set up a
> FreeSwitch on a mac in my office... and I've tried the same experiment
> going to that box... again, silence.
>
> 4. I've tried using other pcap files that come with SIPP... I never hear
> the audio tones play through.
>
>
> I'm kidna at a loss here, and have been trying different things on my own
> for several days. If anyone has any advice, I'd appreciate it.
>
> Ultimately my goal is to have confidence in a functional test that
> automates a scenario:
> - SIPP calls our integration/uat FreeSwitch
> - The call is redirected to a SIPP UAS
> - UAC sends RTP audio
> - UAS uses a rtp_echo
> - I get some packets verifying there's a RTP duration and the test is
> labeled a pass.
>
> But first I'd like to hear the audio actually come through.
>
>
>
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