I need to build a test suite for a service provider's SIP service infrastructure. The box will initiate and accept calls and verify that SIP messages are valid, audio media is two-way, and call control features are also working. My plan for the moment is to use FreeSWITCH and to trigger the outbound calls via ESL, and then use packet capture and Perl modules for parsing and verifying the SIP messages. What are other options? I see that SIPp doesn't look too bad, but I never worked with it. This page lists mostly outdated and closed-source packages, so not very helpful: http://www.voip-info.org/wiki/view/Protocol+Verification+and+Testing I'm planning to publish as much of my work as possible as open source. Any input will be appreciated. thanks, stan
just found the Sippy Cup which looks promising: https://github.com/mojolingo/sippy_cup On Thu, Mar 19, 2015 at 4:43 PM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
I need to build a test suite for a service provider's SIP service infrastructure. The box will initiate and accept calls and verify that SIP messages are valid, audio media is two-way, and call control features are also working.
My plan for the moment is to use FreeSWITCH and to trigger the outbound calls via ESL, and then use packet capture and Perl modules for parsing and verifying the SIP messages.
What are other options? I see that SIPp doesn't look too bad, but I never worked with it.
This page lists mostly outdated and closed-source packages, so not very helpful: http://www.voip-info.org/wiki/view/Protocol+Verification+and+Testing
I'm planning to publish as much of my work as possible as open source.
Any input will be appreciated.
thanks, stan
Stanislav, also look into http://sipp.sourceforge.net/doc/reference.html Thanks, Tito On Thu, Mar 19, 2015 at 11:54 AM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
just found the Sippy Cup which looks promising: https://github.com/mojolingo/sippy_cup
On Thu, Mar 19, 2015 at 4:43 PM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
I need to build a test suite for a service provider's SIP service infrastructure. The box will initiate and accept calls and verify that SIP messages are valid, audio media is two-way, and call control features are also working.
My plan for the moment is to use FreeSWITCH and to trigger the outbound calls via ESL, and then use packet capture and Perl modules for parsing and verifying the SIP messages.
What are other options? I see that SIPp doesn't look too bad, but I never worked with it.
This page lists mostly outdated and closed-source packages, so not very helpful: http://www.voip-info.org/wiki/view/Protocol+Verification+and+Testing
I'm planning to publish as much of my work as possible as open source.
Any input will be appreciated.
thanks, stan
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
sippy_cup actually generates test scenarios that run on sipp. On Thu, Mar 19, 2015 at 12:42 PM, Tito Cumpen <soulofmischief87@gmail.com> wrote:
Stanislav,
also look into http://sipp.sourceforge.net/doc/reference.html
Thanks, Tito
On Thu, Mar 19, 2015 at 11:54 AM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
just found the Sippy Cup which looks promising: https://github.com/mojolingo/sippy_cup
On Thu, Mar 19, 2015 at 4:43 PM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
I need to build a test suite for a service provider's SIP service infrastructure. The box will initiate and accept calls and verify that SIP messages are valid, audio media is two-way, and call control features are also working.
My plan for the moment is to use FreeSWITCH and to trigger the outbound calls via ESL, and then use packet capture and Perl modules for parsing and verifying the SIP messages.
What are other options? I see that SIPp doesn't look too bad, but I never worked with it.
This page lists mostly outdated and closed-source packages, so not very helpful: http://www.voip-info.org/wiki/view/Protocol+Verification+and+Testing
I'm planning to publish as much of my work as possible as open source.
Any input will be appreciated.
thanks, stan
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Stan, I started something on Github a while ago utilizing sipp, sox, wireshark, and pcapsipdump to automate call quality monitoring with Nagios. I was planning on expanding it to monitor some TTS and ASR service but never got the cycle to continue. I believe we all needed some automated monitoring in our infrastructure. I am hoping to get more people involved so we can all benefit from the fruits of our collaboration. The documentation is not complete, but you should be able to tell what I am trying to do in the perl script. https://github.com/bbhenry/check_voip_call Thanks, On Thu, Mar 19, 2015 at 8:43 AM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
I need to build a test suite for a service provider's SIP service infrastructure. The box will initiate and accept calls and verify that SIP messages are valid, audio media is two-way, and call control features are also working.
My plan for the moment is to use FreeSWITCH and to trigger the outbound calls via ESL, and then use packet capture and Perl modules for parsing and verifying the SIP messages.
What are other options? I see that SIPp doesn't look too bad, but I never worked with it.
This page lists mostly outdated and closed-source packages, so not very helpful: http://www.voip-info.org/wiki/view/Protocol+Verification+and+Testing
I'm planning to publish as much of my work as possible as open source.
Any input will be appreciated.
thanks, stan
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (4)
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Christopher Rienzo
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Henry Huang
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Stanislav Sinyagin
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Tito Cumpen