Re: [asterisk-users] [OT] How to use audio files with SIPp
On Fri, Feb 9, 2018 at 8:04 AM, Olivier wrote: > Thank you very much George for replying. > > 2018-02-09 14:39 GMT+01:00 George Joseph : > >> >> >> On Fri, Feb 9, 2018 at 6:27 AM, Olivier 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 > > >> >>> -- >>> _ >>> -- 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 >>> >> >> >> >> >> >> -- >> 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 >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > -- > _ > -- 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 > -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] How to use audio files with SIPp
On Fri, 9 Feb 2018, Olivier wrote: 3. How do you capture an RTP flux with thark or tcpdump ? Take a look at 'pcapsipdump.' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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
Re: [asterisk-users] [OT] How to use audio files with SIPp
Thank you very much George for replying. 2018-02-09 14:39 GMT+01:00 George Joseph : > > > On Fri, Feb 9, 2018 at 6:27 AM, Olivier 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 ? 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. [1] https://stackoverflow.com/questions/20122607/playing-audio-file-using-sipp/20123193 > >> -- >> _ >> -- 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 >> > > > > > > -- > 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 > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- 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
Re: [asterisk-users] [OT] How to use audio files with SIPp
On Fri, Feb 9, 2018 at 6:27 AM, Olivier 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. > > 2. When generating simultaneous calls from one source device to a single > target device, do you need to have specific PCAP files (one specific for > each call) with specific source port ? > Nope. See above. > > 3. How do you capture an RTP flux with thark or tcpdump ? > This is a little tricky. tcpdump isn't much help if you don't know the ports. With tshark though you can create a "read" filter that can capture only RTP packets but it's very expensive. Usually, I just capture all UDP packets between the hosts in question with tcpdump, then use the Wireshark gui to filter the rtp stream. Then you can just export those packets. > > Best regards > > [1] http://sipp.sourceforge.net/doc/reference.html#PCAP+Play > > > -- > _ > -- 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 > -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [OT] How to use audio files with SIPp
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 ? 2. When generating simultaneous calls from one source device to a single target device, do you need to have specific PCAP files (one specific for each call) with specific source port ? 3. How do you capture an RTP flux with thark or tcpdump ? Best regards [1] http://sipp.sourceforge.net/doc/reference.html#PCAP+Play -- _ -- 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
