Hello,
wondering if anyone here is aware of a lightweight sip app that can answer a call, play some file and/or do echo mode, mainly targeted at using it for basic sip routing and call testing. Of course I know that Asterisk and FreeSwitch (or even SEMS) can do that, but they have many dependencies, requiring quite some resources to run them, so I thought maybe someone here figured out different solutions, eventually cli based apps like pjsua or baresip. GUI apps for Linux are also fine if they can be configured for such behaviour.
Cheers, Daniel
Sipp since version 3.3 ( as I remember ) can play files. I have used that for some kind of testing.
On Thu, 26 Mar 2020, 08:45 Daniel-Constantin Mierla, miconda@gmail.com wrote:
Hello,
wondering if anyone here is aware of a lightweight sip app that can answer a call, play some file and/or do echo mode, mainly targeted at using it for basic sip routing and call testing. Of course I know that Asterisk and FreeSwitch (or even SEMS) can do that, but they have many dependencies, requiring quite some resources to run them, so I thought maybe someone here figured out different solutions, eventually cli based apps like pjsua or baresip. GUI apps for Linux are also fine if they can be configured for such behaviour.
Cheers, Daniel
-- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Thanks! The inconvenient thing with sipp is that I have to write the xml scenarios, which can be different if the call goes through many proxies or single proxy or just behind another media server like asterisk or freeswitch.
Cheers, Daniel
On 26.03.20 08:53, Yuriy Gorlichenko wrote:
Sipp since version 3.3 ( as I remember ) can play files. I have used that for some kind of testing.
On Thu, 26 Mar 2020, 08:45 Daniel-Constantin Mierla, <miconda@gmail.com mailto:miconda@gmail.com> wrote:
Hello, wondering if anyone here is aware of a lightweight sip app that can answer a call, play some file and/or do echo mode, mainly targeted at using it for basic sip routing and call testing. Of course I know that Asterisk and FreeSwitch (or even SEMS) can do that, but they have many dependencies, requiring quite some resources to run them, so I thought maybe someone here figured out different solutions, eventually cli based apps like pjsua or baresip. GUI apps for Linux are also fine if they can be configured for such behaviour. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com <http://www.asipto.com> www.twitter.com/miconda <http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda> _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
I’m using pjsua in test suite. It is lightweight, it can be controlled by telnet. It is easily can be built for Alpine and run in the docker. Docker image size is 19 MB. It can play file when calling, it can answer and play file, it can record call. And fully supports IPv4, IPv6, TLS/TCP/UDP, SRTP so we can simulate a lot of cases.
----- Alexey Vasilyev alexei.vasilyev@gmail.com
26 Mar 2020, в 08:43, Daniel-Constantin Mierla miconda@gmail.com написал(а):
Hello,
wondering if anyone here is aware of a lightweight sip app that can answer a call, play some file and/or do echo mode, mainly targeted at using it for basic sip routing and call testing. Of course I know that Asterisk and FreeSwitch (or even SEMS) can do that, but they have many dependencies, requiring quite some resources to run them, so I thought maybe someone here figured out different solutions, eventually cli based apps like pjsua or baresip. GUI apps for Linux are also fine if they can be configured for such behaviour.
Cheers, Daniel
-- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
I looked a bit at the pjsua recently (and used it along the time), but I couldn't find an option to do echo mode. Are you aware if it can be done?
Cheers, Daniel
On 26.03.20 09:08, Alexey Vasilyev wrote:
I’m using pjsua in test suite. It is lightweight, it can be controlled by telnet. It is easily can be built for Alpine and run in the docker. Docker image size is 19 MB. It can play file when calling, it can answer and play file, it can record call. And fully supports IPv4, IPv6, TLS/TCP/UDP, SRTP so we can simulate a lot of cases.
Alexey Vasilyev alexei.vasilyev@gmail.com
26 Mar 2020, в 08:43, Daniel-Constantin Mierla miconda@gmail.com написал(а):
Hello,
wondering if anyone here is aware of a lightweight sip app that can answer a call, play some file and/or do echo mode, mainly targeted at using it for basic sip routing and call testing. Of course I know that Asterisk and FreeSwitch (or even SEMS) can do that, but they have many dependencies, requiring quite some resources to run them, so I thought maybe someone here figured out different solutions, eventually cli based apps like pjsua or baresip. GUI apps for Linux are also fine if they can be configured for such behaviour.
Cheers, Daniel
-- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
baresip is also something to consider https://github.com/alfredh/baresip
Regards,
Nuno
On Thu, Mar 26, 2020 at 8:56 AM Daniel-Constantin Mierla miconda@gmail.com wrote:
I looked a bit at the pjsua recently (and used it along the time), but I couldn't find an option to do echo mode. Are you aware if it can be done?
Cheers, Daniel
On 26.03.20 09:08, Alexey Vasilyev wrote:
I’m using pjsua in test suite. It is lightweight, it can be controlled
by telnet.
It is easily can be built for Alpine and run in the docker. Docker image
size is 19 MB.
It can play file when calling, it can answer and play file, it can
record call.
And fully supports IPv4, IPv6, TLS/TCP/UDP, SRTP so we can simulate a
lot of cases.
Alexey Vasilyev alexei.vasilyev@gmail.com
26 Mar 2020, в 08:43, Daniel-Constantin Mierla miconda@gmail.com
написал(а):
Hello,
wondering if anyone here is aware of a lightweight sip app that can answer a call, play some file and/or do echo mode, mainly targeted at using it for basic sip routing and call testing. Of course I know that Asterisk and FreeSwitch (or even SEMS) can do that, but they have many dependencies, requiring quite some resources to run them, so I thought maybe someone here figured out different solutions, eventually cli based apps like pjsua or baresip. GUI apps for Linux are also fine if they can be configured for such behaviour.
Cheers, Daniel
-- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
On 26.03.20 11:19, Juha Heinanen wrote:
Daniel-Constantin Mierla writes:
wondering if anyone here is aware of a lightweight sip app that can answer a call, play some file and/or do echo mode, mainly targeted at using it for basic sip routing and call testing.
baresip cli app can do all that.
OK, thanks, I will look into it! One more quick question: this is something that can be done directly via command line parameters or its config file?
Asking because in the past, with pjsua I had to write small shell command wrappers to simulate sending commands -- like for example sending the command set in $CMD after 30secs:
(sleep 30; echo $CMD) | pjsua ...
Cheers, Daniel
Hello,
you can give the important stuff (like codecs, play file, auto accept etc..) on the command line.
Cheers,
Henning
Daniel-Constantin Mierla writes:
wondering if anyone here is aware of a lightweight sip app that can answer a call, play some file and/or do echo mode, mainly targeted at using it for basic sip routing and call testing.
baresip cli app can do all that.
OK, thanks, I will look into it! One more quick question: this is something that can be done directly via command line parameters or its config file?
Here is an example configuration of a SIP UA that plays a file when it receives a call:
.baresip/config: ... audio_source gst,file:///tmp/file_to_play.wav ... module gst1.so ...
.baresip/accounts: "Play" sip:play@test.tutpro.com;auth_user="play";auth_pass="something";outbound="sip:192.87.89.90:5060;transport=tcp";ptime=20;audio_codecs=OPUS/48000/2,PCMU,PCMA;regint=600;pubint=0;regq=0.5;sipnat=outbound;answermode=auto
-- Juha
Asking because in the past, with pjsua I had to write small shell command wrappers to simulate sending commands -- like for example sending the command set in $CMD after 30secs:
(sleep 30; echo $CMD) | pjsua ...
Cheers, Daniel
-- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda
Thanks Juha, useful for a quicker starting point -- hope to get a bit time soon to play with and eventually replace some of the heavy apps I use in couple of testing scenarios.
Cheers, Daniel
On 26.03.20 17:31, Juha Heinanen wrote:
Daniel-Constantin Mierla writes:
wondering if anyone here is aware of a lightweight sip app that can answer a call, play some file and/or do echo mode, mainly targeted at using it for basic sip routing and call testing.
baresip cli app can do all that.
OK, thanks, I will look into it! One more quick question: this is something that can be done directly via command line parameters or its config file?
Here is an example configuration of a SIP UA that plays a file when it receives a call:
.baresip/config: ... audio_source gst,file:///tmp/file_to_play.wav ... module gst1.so ...
.baresip/accounts: "Play" sip:play@test.tutpro.com;auth_user="play";auth_pass="something";outbound="sip:192.87.89.90:5060;transport=tcp";ptime=20;audio_codecs=OPUS/48000/2,PCMU,PCMA;regint=600;pubint=0;regq=0.5;sipnat=outbound;answermode=auto
-- Juha
Asking because in the past, with pjsua I had to write small shell command wrappers to simulate sending commands -- like for example sending the command set in $CMD after 30secs:
(sleep 30; echo $CMD) | pjsua ...
Cheers, Daniel
-- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda
Daniel-Constantin Mierla writes:
Here is an example configuration of a SIP UA that plays a file when it receives a call:
.baresip/config: ... audio_source gst,file:///tmp/file_to_play.wav ... module gst1.so
gst1 module was recently renamed to gst. So the last line may need to be replaced by
module gst.so
-- Juha
Regarding echo, one can use same kind of baresip accounts file as with play, such as
sip:echo@test.tutpro.com;auth_user=echo;auth_pass=xxxxxx;outbound="sip:192.168.43.82:5060;transport=tcp";ptime=20;audio_codecs=OPUS/48000/2,PCMU,PCMA;regint=600;pubint=0;regq=0.5;sipnat=outbound;answermode=auto
The difference is that in config file, these lines are needed:
audio_player aubridge,pseudo0 audio_source aubridge,pseudo0 module aubridge.so module_app echo.so
-- Juha
Thanks for this hints as well!
Would it be possible to combine like playing a file first, then do echo mode? What I could figure out from your examples, it is mainly about what modules and module_app's are loaded in baresip config. The config for playing an audio file has:
audio_source gst,file:///tmp/file_to_play.wav
and the one for echo:
audio_source aubridge,pseudo0
So it looks like a conflict to be able to have both.
Having two independent accounts that one plays file and one does echo is enough for what I wanted to find, I am asking more for curiosity.
Cheers, Daniel
On 27.03.20 12:05, Juha Heinanen wrote:
Regarding echo, one can use same kind of baresip accounts file as with play, such as
sip:echo@test.tutpro.com;auth_user=echo;auth_pass=xxxxxx;outbound="sip:192.168.43.82:5060;transport=tcp";ptime=20;audio_codecs=OPUS/48000/2,PCMU,PCMA;regint=600;pubint=0;regq=0.5;sipnat=outbound;answermode=auto
The difference is that in config file, these lines are needed:
audio_player aubridge,pseudo0 audio_source aubridge,pseudo0 module aubridge.so module_app echo.so
-- Juha
Hi Daniel,
A little late, but I created a quick start doc on how to use voip_patrol docker container pre-configured to answer anything and play a file
https://github.com/jchavanton/voip_patrol/blob/quick_start/QUICK_START.md
The docker image is quite large I keep it easily debugable, in this end this is only disk space
On Fri, Mar 27, 2020 at 4:43 AM Daniel-Constantin Mierla miconda@gmail.com wrote:
Thanks for this hints as well!
Would it be possible to combine like playing a file first, then do echo mode? What I could figure out from your examples, it is mainly about what modules and module_app's are loaded in baresip config. The config for playing an audio file has:
audio_source gst,file:///tmp/file_to_play.wav
and the one for echo:
audio_source aubridge,pseudo0
So it looks like a conflict to be able to have both.
Having two independent accounts that one plays file and one does echo is enough for what I wanted to find, I am asking more for curiosity.
Cheers, Daniel
On 27.03.20 12:05, Juha Heinanen wrote:
Regarding echo, one can use same kind of baresip accounts file as with play, such as
<sip:echo@test.tutpro.com ;auth_user=echo;auth_pass=xxxxxx;outbound="sip:192.168.43.82:5060
;transport=tcp";ptime=20;audio_codecs=OPUS/48000/2,PCMU,PCMA;regint=600;pubint=0;regq=0.5;sipnat=outbound;answermode=auto
The difference is that in config file, these lines are needed:
audio_player aubridge,pseudo0 audio_source aubridge,pseudo0 module aubridge.so module_app echo.so
-- Juha
-- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
I also have an image for the RTP media server module, I think is it also coming ready to play a file.
https://hub.docker.com/repository/docker/jchavanton/rtp_media_server
On Tue, Mar 31, 2020 at 10:21 PM Julien Chavanton jchavanton@gmail.com wrote:
Hi Daniel,
A little late, but I created a quick start doc on how to use voip_patrol docker container pre-configured to answer anything and play a file
https://github.com/jchavanton/voip_patrol/blob/quick_start/QUICK_START.md
The docker image is quite large I keep it easily debugable, in this end this is only disk space
On Fri, Mar 27, 2020 at 4:43 AM Daniel-Constantin Mierla < miconda@gmail.com> wrote:
Thanks for this hints as well!
Would it be possible to combine like playing a file first, then do echo mode? What I could figure out from your examples, it is mainly about what modules and module_app's are loaded in baresip config. The config for playing an audio file has:
audio_source gst,file:///tmp/file_to_play.wav
and the one for echo:
audio_source aubridge,pseudo0
So it looks like a conflict to be able to have both.
Having two independent accounts that one plays file and one does echo is enough for what I wanted to find, I am asking more for curiosity.
Cheers, Daniel
On 27.03.20 12:05, Juha Heinanen wrote:
Regarding echo, one can use same kind of baresip accounts file as with play, such as
<sip:echo@test.tutpro.com ;auth_user=echo;auth_pass=xxxxxx;outbound="sip:192.168.43.82:5060
;transport=tcp";ptime=20;audio_codecs=OPUS/48000/2,PCMU,PCMA;regint=600;pubint=0;regq=0.5;sipnat=outbound;answermode=auto
The difference is that in config file, these lines are needed:
audio_player aubridge,pseudo0 audio_source aubridge,pseudo0 module aubridge.so module_app echo.so
-- Juha
-- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hello,
thanks, looks interesting, need to look more at voip_patrol tool. Is the xml language for config/scenarios documented?
Cheers, Daniel
On 01.04.20 07:21, Julien Chavanton wrote:
Hi Daniel,
A little late, but I created a quick start doc on how to use voip_patrol docker container pre-configured to answer anything and play a file
https://github.com/jchavanton/voip_patrol/blob/quick_start/QUICK_START.md
The docker image is quite large I keep it easily debugable, in this end this is only disk space
On Fri, Mar 27, 2020 at 4:43 AM Daniel-Constantin Mierla <miconda@gmail.com mailto:miconda@gmail.com> wrote:
Thanks for this hints as well! Would it be possible to combine like playing a file first, then do echo mode? What I could figure out from your examples, it is mainly about what modules and module_app's are loaded in baresip config. The config for playing an audio file has: audio_source gst,file:///tmp/file_to_play.wav and the one for echo: audio_source aubridge,pseudo0 So it looks like a conflict to be able to have both. Having two independent accounts that one plays file and one does echo is enough for what I wanted to find, I am asking more for curiosity. Cheers, Daniel On 27.03.20 12:05, Juha Heinanen wrote: > Regarding echo, one can use same kind of baresip accounts file as > with play, such as > > <sip:echo@test.tutpro.com <mailto:sip%3Aecho@test.tutpro.com>>;auth_user=echo;auth_pass=xxxxxx;outbound="sip:192.168.43.82:5060;transport=tcp";ptime=20;audio_codecs=OPUS/48000/2,PCMU,PCMA;regint=600;pubint=0;regq=0.5;sipnat=outbound;answermode=auto > > The difference is that in config file, these lines are needed: > > audio_player aubridge,pseudo0 > audio_source aubridge,pseudo0 > module aubridge.so > module_app echo.so > > -- Juha -- Daniel-Constantin Mierla -- www.asipto.com <http://www.asipto.com> www.twitter.com/miconda <http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda> _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
My pleasure,
The xml actions, parameters and examples are documented in the main readme.
https://github.com/jchavanton/voip_patrol/blob/master/README.md
On Wed, Apr 1, 2020 at 1:32 AM Daniel-Constantin Mierla miconda@gmail.com wrote:
Hello,
thanks, looks interesting, need to look more at voip_patrol tool. Is the xml language for config/scenarios documented?
Cheers, Daniel On 01.04.20 07:21, Julien Chavanton wrote:
Hi Daniel,
A little late, but I created a quick start doc on how to use voip_patrol docker container pre-configured to answer anything and play a file
https://github.com/jchavanton/voip_patrol/blob/quick_start/QUICK_START.md
The docker image is quite large I keep it easily debugable, in this end this is only disk space
On Fri, Mar 27, 2020 at 4:43 AM Daniel-Constantin Mierla < miconda@gmail.com> wrote:
Thanks for this hints as well!
Would it be possible to combine like playing a file first, then do echo mode? What I could figure out from your examples, it is mainly about what modules and module_app's are loaded in baresip config. The config for playing an audio file has:
audio_source gst,file:///tmp/file_to_play.wav
and the one for echo:
audio_source aubridge,pseudo0
So it looks like a conflict to be able to have both.
Having two independent accounts that one plays file and one does echo is enough for what I wanted to find, I am asking more for curiosity.
Cheers, Daniel
On 27.03.20 12:05, Juha Heinanen wrote:
Regarding echo, one can use same kind of baresip accounts file as with play, such as
sip:echo@test.tutpro.com;auth_user=echo;auth_pass=xxxxxx;outbound=
"sip:192.168.43.82:5060;transport=tcp" ;ptime=20;audio_codecs=OPUS/48000/2,PCMU,PCMA;regint=600;pubint=0;regq=0.5;sipnat=outbound;answermode=auto
The difference is that in config file, these lines are needed:
audio_player aubridge,pseudo0 audio_source aubridge,pseudo0 module aubridge.so module_app echo.so
-- Juha
-- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda