Re: [sr-dev] [kamailio/kamailio] rtp_media_server: adding module (#1701)

[Henning Westerholt]

Thank you!

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Re: [sr-dev] [kamailio/kamailio] rtp_media_server: adding module (#1701)

[Julien Chavanton]

Merged #1701 into master.

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Re: [sr-dev] [kamailio/kamailio] rtp_media_server: adding module (#1701)

[Julien Chavanton]

I believe everything in the review was addressed

I did remove the audio file and added a link to download one.

I added another commit to support multiple actions this will be needed by some 
action like DTMF capture that could interrupt play and probably many other 
future commands.

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Re: [sr-dev] [kamailio/kamailio] rtp_media_server: adding module (#1701)

[Daniel-Constantin Mierla]

Although I didn't have time to look at the code, I am fine with merging it, 
being a lot of time till 5.3 release that allows testing and tuning, if someone 
is interested.

Actually, I wanted to write here to just say that @jchavanton has write access 
to kamailio already. So he can also merge and commit already.

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Re: [sr-dev] [kamailio/kamailio] rtp_media_server: adding module (#1701)

[Henning Westerholt]

@jchavanton can you remove the WAV file as well? As there were no comments from 
other developers I can then add the module to the master branch. I will grant 
you access to do the further changes directly in the repository.

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Re: [sr-dev] [kamailio/kamailio] rtp_media_server: adding module (#1701)

[Julien Chavanton]

> ohk. Because what I will try with module by converting out of band dtmf to in 
> band dtmf. On Fri, 9 Nov 2018 at 12:28 PM, Julien Chavanton 
>  wrote:
> Just curious to know that rtp_media_server module will not conflict with 
> rtpengine module. On Fri, 9 Nov 2018 at 12:11 PM, Julien Chavanton 
> ***@***.***> wrote: revisited the synchronization and added a few more error 
> handling ... We are sure we want to remove the small voice file ? 157K 
> Bach_10s_8000.wav — You are receiving this because you are subscribed to this 
> thread. Reply to this email directly, view it on GitHub <#1701 (comment) 
> <[#1701 
> (comment)](https://github.com/kamailio/kamailio/pull/1701#issuecomment-437265721)>>,
>  or mute the thread 
> https://github.com/notifications/unsubscribe-auth/AZoKAh3lBVGahBWPHLQkyWSn-BvSAHR1ks5utSOPgaJpZM4YF7gV
>  . -- Surendra Tiwari VoIP Application Developer, Plivo Inc., 201 Mission 
> Street, 
> *Suite
>  230 
> *,
>  San Francisco - 94105, USA 
> 
>  Web: [www.plivo.com](http://www.plivo.com) | Twitter: @plivo 
>  Hi Surendra, RTP engine module is encoding the SDP 
> and transferring it to RTP engine server application. RTP Media Server is 
> starting an RTP enpoint in a thread from a Kamailio forked process. There 
> nothing related that can conflict, expect the fact that they will both play 
> with the SDP, of course you can not use both module on the same call at the 
> same time :) but you could use both module on the same Kamailio 
> instance/config — You are receiving this because you commented. Reply to this 
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> 
>  .
> -- Surendra Tiwari VoIP Application Developer, Plivo Inc., 201 Mission 
> Street, *Suite 230*, San Francisco - 94105, USA Web: 
> [www.plivo.com](http://www.plivo.com) | Twitter: @plivo

Not sure I understand your last comment

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Re: [sr-dev] [kamailio/kamailio] rtp_media_server: adding module (#1701)

[Surendra Tiwari]

ohk. Because what I will try with module by converting out of band dtmf to
in band dtmf.

On Fri, 9 Nov 2018 at 12:28 PM, Julien Chavanton 
wrote:

> Just curious to know that rtp_media_server module will not conflict with
> rtpengine module.
> On Fri, 9 Nov 2018 at 12:11 PM, Julien Chavanton ***@***.***> wrote:
> revisited the synchronization and added a few more error handling ... We
> are sure we want to remove the small voice file ? 157K Bach_10s_8000.wav —
> You are receiving this because you are subscribed to this thread. Reply to
> this email directly, view it on GitHub <#1701 (comment)
> >,
> or mute the thread
> https://github.com/notifications/unsubscribe-auth/AZoKAh3lBVGahBWPHLQkyWSn-BvSAHR1ks5utSOPgaJpZM4YF7gV
> .
> -- Surendra Tiwari VoIP Application Developer, Plivo Inc., 201 Mission
> Street,
> *Suite
> 230
> *,
> San Francisco - 94105, USA
> 
> Web: www.plivo.com | Twitter: @plivo 
>
> Hi Surendra, RTP engine module is encoding the SDP and transferring it to
> RTP engine server application.
>
> RTP Media Server is starting an RTP enpoint in a thread from a Kamailio
> forked process.
>
> There nothing related that can conflict, expect the fact that they will
> both play with the SDP, of course you can not use both module on the same
> call at the same time :) but you could use both module on the same Kamailio
> instance/config
>
> —
> You are receiving this because you commented.
>
>
> Reply to this email directly, view it on GitHub
> ,
> or mute the thread
> 
> .
>
-- 
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VoIP Application Developer,
Plivo Inc., 201 Mission Street, *Suite 230*, San Francisco - 94105, USA
Web: www.plivo.com | Twitter: @plivo


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Re: [sr-dev] [kamailio/kamailio] rtp_media_server: adding module (#1701)

[Julien Chavanton]

> Just curious to know that rtp_media_server module will not conflict with 
> rtpengine module.
> On Fri, 9 Nov 2018 at 12:11 PM, Julien Chavanton ***@***.***> wrote: 
> revisited the synchronization and added a few more error handling ... We are 
> sure we want to remove the small voice file ? 157K Bach_10s_8000.wav — You 
> are receiving this because you are subscribed to this thread. Reply to this 
> email directly, view it on GitHub <[#1701 
> (comment)](https://github.com/kamailio/kamailio/pull/1701#issuecomment-437265721)>,
>  or mute the thread 
> 
>  .
> -- Surendra Tiwari VoIP Application Developer, Plivo Inc., 201 Mission 
> Street, *Suite 230*, San Francisco - 94105, USA Web: 
> [www.plivo.com](http://www.plivo.com) | Twitter: @plivo

Hi Surendra, RTP engine module is encoding the SDP and transferring it to RTP 
engine server application.

RTP Media Server is starting an RTP enpoint in a thread from a Kamailio forked 
process.

There nothing related that can conflict, expect the fact that they will both 
play with the SDP, of course you can not use both module on the same call at 
the same time :) but you could use both module on the same Kamailio 
instance/config 


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Re: [sr-dev] [kamailio/kamailio] rtp_media_server: adding module (#1701)

[Henning Westerholt]

> We are sure we want to remove the small voice file ? 157K Bach_10s_8000.wav

I would just adding a link in the README to e.g. this, they provide voice files 
in different languages:
http://www.voiptroubleshooter.com/open_speech/


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Re: [sr-dev] [kamailio/kamailio] rtp_media_server: adding module (#1701)

[Surendra Tiwari]

Just curious to know that rtp_media_server module will not conflict with
rtpengine module.

On Fri, 9 Nov 2018 at 12:11 PM, Julien Chavanton 
wrote:

> revisited the synchronization and added a few more error handling ...
> We are sure we want to remove the small voice file ? 157K
> Bach_10s_8000.wav
>
> —
> You are receiving this because you are subscribed to this thread.
> Reply to this email directly, view it on GitHub
> ,
> or mute the thread
> 
> .
>
-- 
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VoIP Application Developer,
Plivo Inc., 201 Mission Street, *Suite 230*, San Francisco - 94105, USA
Web: www.plivo.com | Twitter: @plivo


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Re: [sr-dev] [kamailio/kamailio] rtp_media_server: adding module (#1701)

[Julien Chavanton]

revisited the synchronization and added a few more error handling ...
We are sure we want to remove  the small voice file ? `157K Bach_10s_8000.wav`



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Re: [sr-dev] [kamailio/kamailio] rtp_media_server: adding module (#1701)

[Julien Chavanton]

@jchavanton pushed 1 commit.

78d1090  rtp_media_server: review part2 (will be squashed)


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Re: [sr-dev] [kamailio/kamailio] rtp_media_server: adding module (#1701)

[Julien Chavanton]

Thanks, this is good news !

So far there was only a brief discussion in slack, but nothing valuable enough 
to share in the mailing list.
Mainly I shared my optimism about the fact that these libs are also designed to 
build server side components.
The very minimalist feature set currently included was questioned, I believe 
that extending the features is not such a big endeavor considering the 
architecture of the routing script and the libraries. 

I still need to address some comments you have made.


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Re: [sr-dev] [kamailio/kamailio] rtp_media_server: adding module (#1701)

[Henning Westerholt]

Hello Julien, thank you for updating the pull request. I also saw your message 
on sr-dev, so far no reply from the other developers. About the code - I will 
have another closer look on Friday. if there are no objections by then from the 
list we can discuss about integrating it into the master branch.

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Re: [sr-dev] [kamailio/kamailio] rtp_media_server: adding module (#1701)

[Julien Chavanton]

jchavanton commented on this pull request.



> + char sdp_o[128];
+   snprintf(
+   sdp_o, 128, "o=- 1028316687 1 IN IP4 %s\r\n", 
sdp_info->local_ip.s);
+   body->len += strlen(sdp_o);
+
+   // (connection information -- not required if included in all media)
+   char sdp_c[128];
+   snprintf(sdp_c, 128, "c=IN IP4 %s\r\n", sdp_info->local_ip.s);
+   body->len += strlen(sdp_c);
+
+   char sdp_m[128];
+   snprintf(sdp_m, 128, "m=audio %d RTP/AVP %d\r\n", 
sdp_info->udp_local_port,
+   payload_type_number);
+   body->len += strlen(sdp_m);
+
+   body->s = pkg_malloc(body->len + 1);

check that none was left unchecked

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Re: [sr-dev] [kamailio/kamailio] rtp_media_server: adding module (#1701)

[Julien Chavanton]

jchavanton commented on this pull request.



> +{
+   if(rank == PROC_MAIN) {
+   int pid;
+   pid = fork_process(PROC_XWORKER, "RTP_media_server", 1);
+   if(pid < 0)
+   return -1;
+   if(pid == 0) {
+   rms_session_manage_loop();
+   return 0;
+   }
+   }
+   int rtn = 0;
+   return (rtn);
+}
+
+int rms_str_dup(str *dst, str *src, int shared)

Same topic as before, since they do not convert to `\0` terminated `char *` 
they are not flexible in this case.

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Re: [sr-dev] [kamailio/kamailio] rtp_media_server: adding module (#1701)

[Julien Chavanton]

@jchavanton pushed 1 commit.

592b6d5  rtp_media_server: review part1 (will be squashed)


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Re: [sr-dev] [kamailio/kamailio] rtp_media_server: adding module (#1701)

[Julien Chavanton]

jchavanton commented on this pull request.



> +
+// https://tools.ietf.org/html/rfc4566
+// (protocol version)
+const char *sdp_v = "v=0\r\n";
+// (session name)
+const char *sdp_s = "s=-\r\n";
+// (time the session is active)
+const char *sdp_t = "t=0 0\r\n";
+//"a=rtpmap:101 telephone-event/8000\r\n"
+//"a=fmtp:101 0-15\r\n";
+//"a=rtpmap:0 PCMU/8000\r\n"
+//"a=rtpmap:8 PCMA/8000\r\n"
+//"a=rtpmap:96 opus/48000/2\r\n"
+//"a=fmtp:96 useinbandfec=1\r\n";
+
+static char *rms_shm_strdup(char *source)

I just did refactoring for now , we can clarify if it is best to use helper 
function from the core, add new ones or not ... 

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Re: [sr-dev] [kamailio/kamailio] rtp_media_server: adding module (#1701)

[Julien Chavanton]

jchavanton commented on this pull request.



> +
+// https://tools.ietf.org/html/rfc4566
+// (protocol version)
+const char *sdp_v = "v=0\r\n";
+// (session name)
+const char *sdp_s = "s=-\r\n";
+// (time the session is active)
+const char *sdp_t = "t=0 0\r\n";
+//"a=rtpmap:101 telephone-event/8000\r\n"
+//"a=fmtp:101 0-15\r\n";
+//"a=rtpmap:0 PCMU/8000\r\n"
+//"a=rtpmap:8 PCMA/8000\r\n"
+//"a=rtpmap:96 opus/48000/2\r\n"
+//"a=fmtp:96 useinbandfec=1\r\n";
+
+static char *rms_shm_strdup(char *source)

Maybe we could add another helper function in the core ?

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Re: [sr-dev] [kamailio/kamailio] rtp_media_server: adding module (#1701)

[Julien Chavanton]

jchavanton commented on this pull request.



> +
+// https://tools.ietf.org/html/rfc4566
+// (protocol version)
+const char *sdp_v = "v=0\r\n";
+// (session name)
+const char *sdp_s = "s=-\r\n";
+// (time the session is active)
+const char *sdp_t = "t=0 0\r\n";
+//"a=rtpmap:101 telephone-event/8000\r\n"
+//"a=fmtp:101 0-15\r\n";
+//"a=rtpmap:0 PCMU/8000\r\n"
+//"a=rtpmap:8 PCMA/8000\r\n"
+//"a=rtpmap:96 opus/48000/2\r\n"
+//"a=fmtp:96 useinbandfec=1\r\n";
+
+static char *rms_shm_strdup(char *source)

The reason is that shm_str_dup does not garanty terminated 'char*', this is one 
less thing to worry about. The module is not trying to optimize string like 
Kamailio is when parsing messages.

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Re: [sr-dev] [kamailio/kamailio] rtp_media_server: adding module (#1701)

[Julien Chavanton]

jchavanton commented on this pull request.



> +#define rms_media_h
+
+#include "../../core/mem/shm.h"
+#include "mediastreamer2/mediastream.h"
+#include "mediastreamer2/msrtp.h"
+#include "mediastreamer2/mediastream.h"
+#include "mediastreamer2/dtmfgen.h"
+#include "mediastreamer2/msfileplayer.h"
+#include "mediastreamer2/msfilerec.h"
+#include "mediastreamer2/msrtp.h"
+#include "mediastreamer2/mstonedetector.h"
+#include "mediastreamer2/msfilter.h"
+//#include 
+#include 
+#include 
+#define MS_UNUSED(x) ((void)(x))

removed, since we now include mscommon.h

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Re: [sr-dev] [kamailio/kamailio] rtp_media_server: adding module (#1701)

[Julien Chavanton]

jchavanton commented on this pull request.



> + }
+   ms_factory_init_plugins(f);
+   ms_factory_enable_statistics(f, TRUE);
+   ms_factory_reset_statistics(f);
+   return f;
+}
+
+int rms_media_init()
+{
+   OrtpMemoryFunctions ortp_memory_functions;
+   ortp_memory_functions.malloc_fun = ptr_shm_malloc;
+   ortp_memory_functions.realloc_fun = ptr_shm_realloc;
+   ortp_memory_functions.free_fun = ptr_shm_free;
+   ortp_set_memory_functions(_memory_functions);
+   ortp_init();
+   vars = ortp_malloc(sizeof(shared_global_vars_t));

no good reason, ortp_malloc is using shm_malloc, I replaced the 2 occurrences 
with shm_malloc for clarity from Kamailio module context 

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Re: [sr-dev] [kamailio/kamailio] rtp_media_server: adding module (#1701)

[Julien Chavanton]

jchavanton commented on this pull request.



> + ms_connection_helper_start();
+   ms_connection_helper_link(, m1->ms_rtprecv, -1, 0);
+   ms_connection_helper_link(, m2->ms_rtpsend, 0, -1);
+
+   // direction 2
+   ms_connection_helper_start();
+   ms_connection_helper_link(, m2->ms_rtprecv, -1, 0);
+   ms_connection_helper_link(, m1->ms_rtpsend, 0, -1);
+
+   ms_ticker_attach_multiple(
+   m1->ms_ticker, m1->ms_rtprecv, m2->ms_rtprecv, NULL);
+
+   return 1;
+}
+
+#define MS_UNUSED(x) ((void)(x))

I modified the includes for mediastreamer2 and included the file that was 
defining this macro instead.


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Re: [sr-dev] [kamailio/kamailio] rtp_media_server: adding module (#1701)

[Julien Chavanton]

Hi Henning, thank you for taking  some time to do the a review,  I will address 
all the comments shortly.

Yes integrating a RTP endpoint directly in Kamailio may seems strange at first.

I did discuss this approach 2 years ago at Kamailio World and we discussed 
about this year at Cluecon with Daniel and Fred, the last feedback I got was 
that this was interesting.

By doing integration with a library we can benefit from the scripting engine 
and probably KEMI.
This does not means that everyone should run this module on a Kamailio sever 
acting as a load balancer, of course CPU intensive operations like resampling 
and encoding may be taking place on the server.

I think the comments in this MR are hiting ser-dev mainling list but it make 
sense to write an email as well.

Much appreciated !


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Re: [sr-dev] [kamailio/kamailio] rtp_media_server: adding module (#1701)

[Henning Westerholt]

Thank you Julien for the contribution of this module. I will do a quick first 
review of the code.

The git master is currently in a feature freeze in preparation for the upcoming 
5.2 release, so no new features can be integrated. So I would suggest to finish 
the mentioned remaining steps in your branch.

Did I understand the purpose of the module correctly, as its implements a RTP 
endpoint directly in Kamailio? This is kind of a new approach (for Kamailio) I 
think, the existing RTP handling modules only place the control channel into 
kamailio, and handle the RTP processing in a separate daemon. 

It would suggest to write an e-mail to sr-dev, describing your module, the 
approach and getting feedback.

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