Audio test do work with SIPp. If you are using SIPp to call a real phone
and you can't hear the DTMF or audio you are playing then it is very likely
that something is wrong with your test scenario. Things you should take a
look at:
- Make sure, codecs are fine and that your audio or telephone-events match
the codec you are negotiating, it's best to leave only this codec in the
SIPp XML script, so the other party cannot change it.
- Put the volume of your real phone in the max (seriously, once I lost
some time searching why the audio was not coming but it was only a volume
problem, the recording was too low...)
- Check the ports and IPs negotiated and used for media streams.
- If you recorded a sound, make sure you saved the correct packets in the
pcap file.
Once I did have some problems with the rtp_echo feature and since then I
haven't used it anymore. But your complain is not only in this case, so
maybe there is some problem with your scenario. If you send wireshark
traces of one call, it will be a lot easier to discover what is going on.
On Sat, Feb 9, 2013 at 12:23 AM, David Luu <manga...@gmail.com> wrote:
> 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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>
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