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app_rtsp_sip's Issues

Compilation Error Asterisk 18.14 on Raspberry pi 4

Hello, when I try to compile Asterisk 18.14 with your module I get the following errors:

[CC] app_rtsp_sip.c -> app_rtsp_sip.o
In file included from /home/openhabian/Downloads/asterisk-18.14.0/include/pj/types.h:33,
                 from /home/openhabian/Downloads/asterisk-18.14.0/include/pjsip/sip_config.h:27,
                 from /home/openhabian/Downloads/asterisk-18.14.0/include/pjsip/sip_types.h:34,
                 from /home/openhabian/Downloads/asterisk-18.14.0/include/pjsip.h:24,
                 from /home/openhabian/Downloads/asterisk-18.14.0/include/pjsua-lib/pjsua.h:30,
                 from app_rtsp_sip.c:88:
/home/openhabian/Downloads/asterisk-18.14.0/include/pj/config.h:264:6: error: #error Endianness must be declared for this processor
 #    error Endianness must be declared for this processor
      ^~~~~
In file included from /home/openhabian/Downloads/asterisk-18.14.0/include/pj/types.h:33,
                 from /home/openhabian/Downloads/asterisk-18.14.0/include/pjsip/sip_config.h:27,
                 from /home/openhabian/Downloads/asterisk-18.14.0/include/pjsip/sip_types.h:34,
                 from /home/openhabian/Downloads/asterisk-18.14.0/include/pjsip.h:24,
                 from /home/openhabian/Downloads/asterisk-18.14.0/include/pjsua-lib/pjsua.h:30,
                 from app_rtsp_sip.c:88:
/home/openhabian/Downloads/asterisk-18.14.0/include/pj/config.h:1377:4: error: #error "PJ_IS_LITTLE_ENDIAN is not defined!"
 #  error "PJ_IS_LITTLE_ENDIAN is not defined!"
    ^~~~~
/home/openhabian/Downloads/asterisk-18.14.0/include/pj/config.h:1381:4: error: #error "PJ_IS_BIG_ENDIAN is not defined!"
 #  error "PJ_IS_BIG_ENDIAN is not defined!"
    ^~~~~
In file included from /home/openhabian/Downloads/asterisk-18.14.0/include/pj/limits.h:28,
                 from /home/openhabian/Downloads/asterisk-18.14.0/include/pj/types.h:34,
                 from /home/openhabian/Downloads/asterisk-18.14.0/include/pjsip/sip_config.h:27,
                 from /home/openhabian/Downloads/asterisk-18.14.0/include/pjsip/sip_types.h:34,
                 from /home/openhabian/Downloads/asterisk-18.14.0/include/pjsip.h:24,
                 from /home/openhabian/Downloads/asterisk-18.14.0/include/pjsua-lib/pjsua.h:30,
                 from app_rtsp_sip.c:88:
/home/openhabian/Downloads/asterisk-18.14.0/include/pj/compat/limits.h:39:4: warning: #warning "limits.h is not found or not supported. LONG_MIN and LONG_MAX " "will be defined by the library in pj/compats/limits.h and " "overridable in config_site.h" [-Wcpp]
 #  warning "limits.h is not found or not supported. LONG_MIN and LONG_MAX " \
    ^~~~~~~
In file included from app_rtsp_sip.c:82:
/home/openhabian/Downloads/asterisk-18.14.0/include/asterisk/module.h:565:22: fatal error: opening dependency file .app_rtsp_sip.o.d: Permission denied
  static const struct ast_module_info *ast_module_info = &__mod_info
                      ^~~~~~~~~~~~~~~
app_rtsp_sip.c:4224:1: note: in expansion of macro ‘AST_MODULE_INFO’
 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "RTSP-SIP Application",
 ^~~~~~~~~~~~~~~
compilation terminated.
make[1]: *** [/home/openhabian/Downloads/asterisk-18.14.0/Makefile.rules:165: app_rtsp_sip.o] Error 1
make: *** [Makefile:396: apps] Error 2

How to solve this problem?

Thank you

Some questions :-)

Hey @tommyjlong
I have succeeded now adding an RTSP stream to the call, leys say you have a SIP doorbell (6000) without any video capabilities
With dialplan below, when 6000 calls 666 (conference) , 6001 , 6002 , 6003, ... are called too to join the conference
When 6001 or 6002 or 6003 picksup, those users will see the RTSP stream as video , because 6000 doesnt have video ... , and "doorbell" already joined the conference..

exten => 666,1,NoOp()
 same => n,Progress()
 same => n,Originate(Local/doorbell@default,exten,default,888,1,,aC(ulaw,alaw,h264)c(6000)n(Doorbell))
 same => n,Originate(PJSIP/6001,exten,default,777,1,,aC(ulaw,alaw,h264)c(6000)n(Doorbell)) 
 same => n,ConfBridge(1,myconferenceroom,default_user)
 same => n,Hangup()
 
exten => 777,1,NoOp()
 same => n,ConfBridge(1,myconferenceroom,admin_user)

exten => 888,1,NoOp()
 same => n,ConfBridge(1,myconferenceroom,default_user)
 
exten => doorbell,1,Answer()
same => n,RTSP-SIP(rtsp://admin:[email protected]:554/Streaming/Channels/102,0,Hikvision,5060)

Above is wotking 'ok' ...

But i want to have more features... is below possible?
Above is based on confernce, it has some disadvantages offcourse, leys say i dont want to use conference

  1. When 6000 calls => 6001, is it possible to inject that RTSP stream as video in the SDP/RTP session? maybe also as early media 183 session, maybe with some customizing with ffmpeg and rtp/sdp ? maybe with modifiyn the dial app.c ? or some easier way ?

  2. if above is not possible, and i do need to use conference, can i use the RTSP/h264 video in an early media stage, when i do an originate to 6001 , maybe with customizing the originate.c app ? Changing the SDP/RTP (ffmpeg) and save the rtsp to a file that will be used?

  3. Above conference is wotking verry good with my softphone (6001), i tried a few (linphone desktop/zoiper/acrobit) i always see the RTSP video ... BUT, when i use Linphone on android, seems i have a black screen instead of the video ... and actually i want to use Linphone as a client, and seems that one is the one bugging me?? Do you have any idea? is it maybe because a framerate/resolution that you used in your app ? Have you tried linphone (android) too? Any idea where i can look at ? i have tried several options in linphone, like codec / video size / fps ... but no help
    Is there something i can try/change in your app/code ?

thnx , much appreciated!!

ERROR[630][C-00000001]: app_rtsp_sip.c:2748 main_loop: -No Authenticate header found

Hey @tommyjlong ,

Can you give me a quick help, i was testing your app, i was able to compile it succesfull, the .so loads....
But it gives me error:

[Apr 27 16:28:41] ERROR[630][C-00000001]: app_rtsp_sip.c:2748 main_loop: -No Authenticate header found

I think the .sh script doesnt work, in your .sh file all the files listed there, i dont see them when i extract asterisk

Actually the complete "source" dir is missing there?

image

What am i missing?
I'm using this Asterisk-Addon for HA, in dockerfile there is also:

    ./configure \
        --with-pjproject-bundled \

https://github.com/TECH7Fox/asterisk-hass-addons/blob/main/asterisk/Dockerfile

I'm testing my own local version , if it works, we will merge this new app of yours, added this to dockerfile:

## Install app_rtsp_sip
WORKDIR /usr/src/asterisk
COPY patches/rtsp_sip_links.sh /usr/src/asterisk
RUN ./rtsp_sip_links.sh
COPY patches/app_rtsp_sip.c /usr/src/asterisk/apps

thnx, appreciated!!

Other questions : peer rtp port is not provided

Hi @tommyjlong

Back testing your app...
Can you tell me more about these warnings below?

[Aug 6 13:42:36] NOTICE[445][C-00000006]: app_rtsp_sip.c:2432 main_loop: >rtsp-sip main loop [Aug 6 13:42:36] WARNING[445][C-00000006]: app_rtsp_sip.c:1882 CreateSDP: peer rtp port is not provided [Aug 6 13:42:57] ERROR[445][C-00000006]: app_rtsp_sip.c:2582 main_loop: ast_read() failed. Bail out!

Thnx

Q: Android app that supports rtsp stream

Hi, question...

I'm looking for an intercom solutiion... my doorstation sends early media in 183 session, so if i call to a softphone, i can see video before i pickup... thats good , but early media (video) doesnt work on ringgroups

So including an RTSP stream is a good alternative... But what Android softphones do support RTSP streams?

thnx in advance

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