[asterisk-users] [OT] How to use audio files with SIPp

[George Joseph]

On Fri, Feb 9, 2018 at 8:04 AM, Olivier <oza.4h07 at gmail.com> wrote:

> Thank you very much George for replying.
>
> 2018-02-09 14:39 GMT+01:00 George Joseph <gjoseph at digium.com>:
>
>>
>>
>> On Fri, Feb 9, 2018 at 6:27 AM, Olivier <oza.4h07 at gmail.com> wrote:
>>
>>> Hello,
>>>
>>> SIPp's PCAP play feature can replay pre-recorded audio stream towards
>>> destination (see [1]).
>>> Doc mentions tcpdump and Wireshark as tools to record such RTP streams
>>> without further details.
>>>
>>>
>>> Looking at SIPp 3.2 source archive, I found PCAP samples in a pcap/
>>> directory.
>>> Sample pcap/g711a.pcap file includes RTP from 10.1.3.1:5000 to
>>> 10.1.6.18:2006
>>>
>>> 1. How can you "forge" IPs and/or ports of a pcap file ?
>>>
>>
>> You don't have to.  sipp only takes the rtp payload from the packets in
>> the pcap then just sends the datagrams to the remote in the scenario.
>>
>
> That is exactly what I'm after !
>
> Before diving into this, can I ask which SIPp version and feature are we
> talking about here ?
>

We're on 3.5.


>
> Since I posted my question, I've read this [1] thread mentionning a new
> WAV file playing capability but this feature required SIPp 3.4 and above.
> On Debian Stetch I'm playing with, packaged SIPp is 3.2.
>

That's pretty old.  I'd recommend compiling from source yourself.  It's
very easy to build.


>
>
> [1] https://stackoverflow.com/questions/20122607/playing-
> audio-file-using-sipp/20123193
>
>
>>
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>>
>>
>>
>>
>>
>> --
>> George Joseph
>> Digium, Inc. | Software Developer
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>> Check us out at: www.digium.com & www.asterisk.org
>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
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>>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
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-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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