Thanks Greg.
I'm wondering if the tests mentioned on SIPP's main site actually were
tested with hearing the tones and audio. I'm not seeing any result that
works. I see the packets get through... but they are blank. Even the
embedded uac_pcap scenario doesn't transmit. i dont get it. But it seems
that would be a basic premise to testing SIP... But then the tool is more
geared to load, and generating lots of RTP packets, even if it has no valid
audio, maybe that's enough for most out there.
I just tried the built in dtmf, and like you, no tone was picked up.
So Greg for your testing, what did you do? did you use a different tool?
On Fri, Feb 8, 2013 at 1:16 PM, Greg Thomas <greg.d.tho...@gmail.com> wrote:
> 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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