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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