Thought I'd add some comments. (resending, as forgot to do reply all 1st
time around)
Audio transmission to/from work with SIPP. We had 2 developers work on it
before. You can play back a PCAP file from SIPP to the other end (to
manually listen for the audio). And as I recall, you can also record audio
with SIPP (or maybe 3rd party tool) as PCAP file for whatever the other end
sent, to which you can then manually verify or run against diff tools or
audio detection tools.
But in our case, I believe we never used the default UAC, UAS scenarios and
custom built our XML scenarios to match the SIP messaging specific to our
company's PBX. So we had like up to 3 XML files, one to register the SIP
extension, one to make the call, and one to answer call. And for
make/answer call, the script would either record PCAP audio or send a PCAP
file.
What you want to do should be achievable. Unfortunately, it may take good
knowledge of SIP protocol messaging, Wireshark sniffing, and mucking with
XML files to be able tweak your solution to work. It took our SIP
developers a while to get a workable demo. And unfortunately, it was so
time consuming, it never got fine tuned & deployed for actual automated
testing. Because company deemed it wasn't worth the ROI to pursue (yet
anyways). If only it was simpler to do. Apparently SIP is a lot harder to
work & interoperate with than say HTTP.
On Fri, Feb 8, 2013 at 4:55 PM, Brian Warner <continuou...@gmail.com> wrote:
> 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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