Streaming out of conference via rtmp
In response to the thread " $500 bounty RTMP to conference room", try something like this: <action application="record_session" data="{modname=mod_av,vw=320,vh=240}rtmp://${ipOfDestination}:${portOfDestination}/${queryPathForDestination}"/>
On Wed, Oct 14, 2020 at 10:43 PM David P <davidswalkabout@gmail.com> wrote:
In response to the thread " $500 bounty RTMP to conference room", try something like this:
<action application="record_session" data="{modname=mod_av,vw=320,vh=240}rtmp://${ipOfDestination}:${portOfDestination}/${queryPathForDestination}"/>
So I am not trying to send rtmp, I am trying to do I think what mod_rtmp was doing and LISTEN for rtmp that is broadcast to the FreeSWITCH server. -Nathan
What are you trying to accomplish exactly? On Thu, 15 Oct 2020 at 15:13, Nathan Stratton <nathan@robotics.net> wrote:
On Wed, Oct 14, 2020 at 10:43 PM David P <davidswalkabout@gmail.com> wrote:
In response to the thread "
$500 bounty RTMP to conference room", try something like this:
<action application="record_session" data="{modname=mod_av,vw=320,vh=240}rtmp://${ipOfDestination}:${portOfDestination}/${queryPathForDestination}"/>
So I am not trying to send rtmp, I am trying to do I think what mod_rtmp was doing and LISTEN for rtmp that is broadcast to the FreeSWITCH server.
-Nathan
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.
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-- Regards, David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
Sure, so we have a number of users that dial into a room, say 4600, they receive audio and video from the moderator but don't send anything themselves because of mute|vmute. My customer wants to instead of calling in as a moderator, to instead send content to the room via SlingStudio like they do with Facebook and Youtube. Sling Studio sends a stream as rtmp://{my ip}/{room} I was hoping I could do that with mod_rtmp but was not able to make it work. So I am looking for another way to send video from the RTMP server to the FreeSWITCH room. I have found SIP to RTMP software, but I am trying to do the opposite, I need to get from RTMP to SIP room. <extension name="Simple add-caller conf: member"> <condition field="destination_number" expression="^(46\d{2})$"> <action application="answer"/> <action application="conference" data="$1@default+flags{mute|vmute}"/> </condition> </extension> <extension name="Simple add-caller conf: moderator"> <condition field="destination_number" expression="^\*(46\d{2}$)"> <action application="answer"/> <action application="digit_action_set_realm" data="moderator"/> <action application="conference" data="$1@simple+flags{moderator}"/> </condition> </extension>
<> nathan stratton
On Thu, Oct 15, 2020 at 10:43 AM David Villasmil < david.villasmil.work@gmail.com> wrote:
What are you trying to accomplish exactly?
On Thu, 15 Oct 2020 at 15:13, Nathan Stratton <nathan@robotics.net> wrote:
On Wed, Oct 14, 2020 at 10:43 PM David P <davidswalkabout@gmail.com> wrote:
In response to the thread "
$500 bounty RTMP to conference room", try something like this:
<action application="record_session" data="{modname=mod_av,vw=320,vh=240}rtmp://${ipOfDestination}:${portOfDestination}/${queryPathForDestination}"/>
So I am not trying to send rtmp, I am trying to do I think what mod_rtmp was doing and LISTEN for rtmp that is broadcast to the FreeSWITCH server.
-Nathan
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services
sales@freeswitch.com
Official FreeSWITCH Sites
https://freeswitch.org/confluence
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
-- Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
There’s a mod_vlc, never used it but might be what you’re looking for. Have you seen it? On Thu, 15 Oct 2020 at 16:41, Nathan Stratton <nathan@robotics.net> wrote:
Sure, so we have a number of users that dial into a room, say 4600, they receive audio and video from the moderator but don't send anything themselves because of mute|vmute. My customer wants to instead of calling in as a moderator, to instead send content to the room via SlingStudio like they do with Facebook and Youtube. Sling Studio sends a stream as rtmp://{my ip}/{room} I was hoping I could do that with mod_rtmp but was not able to make it work. So I am looking for another way to send video from the RTMP server to the FreeSWITCH room. I have found SIP to RTMP software, but I am trying to do the opposite, I need to get from RTMP to SIP room.
<extension name="Simple add-caller conf: member"> <condition field="destination_number" expression="^(46\d{2})$"> <action application="answer"/> <action application="conference" data="$1@default +flags{mute|vmute}"/> </condition> </extension>
<extension name="Simple add-caller conf: moderator"> <condition field="destination_number" expression="^\*(46\d{2}$)"> <action application="answer"/> <action application="digit_action_set_realm" data="moderator"/> <action application="conference" data="$1@simple +flags{moderator}"/> </condition> </extension>
<>
nathan stratton
On Thu, Oct 15, 2020 at 10:43 AM David Villasmil < david.villasmil.work@gmail.com> wrote:
What are you trying to accomplish exactly?
On Thu, 15 Oct 2020 at 15:13, Nathan Stratton <nathan@robotics.net> wrote:
On Wed, Oct 14, 2020 at 10:43 PM David P <davidswalkabout@gmail.com> wrote:
In response to the thread "
$500 bounty RTMP to conference room", try something like this:
<action application="record_session" data="{modname=mod_av,vw=320,vh=240}rtmp://${ipOfDestination}:${portOfDestination}/${queryPathForDestination}"/>
So I am not trying to send rtmp, I am trying to do I think what mod_rtmp was doing and LISTEN for rtmp that is broadcast to the FreeSWITCH server.
-Nathan
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services
sales@freeswitch.com
Official FreeSWITCH Sites
https://freeswitch.org/confluence
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
-- Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services
sales@freeswitch.com
Official FreeSWITCH Sites
https://freeswitch.org/confluence
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
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services
sales@freeswitch.com
Official FreeSWITCH Sites
https://freeswitch.org/confluence
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
-- Regards, David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
Mod_rtmp was intended as a softphone endpoint written in flash. dont use it. litterally load mod_av then via esl, fs_cli etc “conference <confname> [play|record] rtmp://{rest/of/url}” its really that simple. play will pkay the rtmp stream into the conference like any other media file. record will shoot a media stream out to what ever rtmp reflector you are using auch as youtube or facebook or your own private rtmp server. Sent from my iPhone
On Oct 15, 2020, at 08:42, Nathan Stratton <nathan@robotics.net> wrote:
On Wed, Oct 14, 2020 at 10:43 PM David P <davidswalkabout@gmail.com> wrote:
In response to the thread " $500 bounty RTMP to conference room", try something like this:
<action application="record_session" data="{modname=mod_av,vw=320,vh=240}rtmp://${ipOfDestination}:${portOfDestination}/${queryPathForDestination}"/>
So I am not trying to send rtmp, I am trying to do I think what mod_rtmp was doing and LISTEN for rtmp that is broadcast to the FreeSWITCH server.
-Nathan _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
Thanks, so yes, I want to play not record, but when I try to play it's looking for a file, it's not listening on port 1935 for a RTMP stream from the server. I need to ACCEPT a RTMP steam like mod_rtmp did, I understand why mod_rtmp is not working but is there any other way to listen from RTMP on port 1935 and send it into a conference? freeswitch@troc.staging.vocinity.com> conference 4600 play rtmp://localhost -ERR (play) File: rtmp://localhost not found. 2020-10-15 12:38:12.636530 [ERR] avformat.c:1603 Invalid Format So the issue is that play rtmp:// does not listen for an incoming RTMP stream.
<> nathan stratton
On Thu, Oct 15, 2020 at 12:32 PM Ken Rice <krice@freeswitch.org> wrote:
Mod_rtmp was intended as a softphone endpoint written in flash. dont use it. litterally load mod_av then via esl, fs_cli etc “conference <confname> [play|record] rtmp://{rest/of/url}”
its really that simple. play will pkay the rtmp stream into the conference like any other media file. record will shoot a media stream out to what ever rtmp reflector you are using auch as youtube or facebook or your own private rtmp server.
Sent from my iPhone
On Oct 15, 2020, at 08:42, Nathan Stratton <nathan@robotics.net> wrote:
On Wed, Oct 14, 2020 at 10:43 PM David P <davidswalkabout@gmail.com> wrote:
In response to the thread " $500 bounty RTMP to conference room", try something like this:
<action application="record_session" data="{modname=mod_av,vw=320,vh=240}rtmp://${ipOfDestination}:${portOfDestination}/${queryPathForDestination}"/>
So I am not trying to send rtmp, I am trying to do I think what mod_rtmp was doing and LISTEN for rtmp that is broadcast to the FreeSWITCH server.
-Nathan _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
these is to my knowledge no way to listen for a stream to randomly start up. This would require designing a new module to handle the listening service. Sent from my iPhone
On Oct 15, 2020, at 11:43, Nathan Stratton <nathan@robotics.net> wrote:
Thanks, so yes, I want to play not record, but when I try to play it's looking for a file, it's not listening on port 1935 for a RTMP stream from the server. I need to ACCEPT a RTMP steam like mod_rtmp did, I understand why mod_rtmp is not working but is there any other way to listen from RTMP on port 1935 and send it into a conference?
freeswitch@troc.staging.vocinity.com> conference 4600 play rtmp://localhost -ERR (play) File: rtmp://localhost not found.
2020-10-15 12:38:12.636530 [ERR] avformat.c:1603 Invalid Format
So the issue is that play rtmp:// does not listen for an incoming RTMP stream.
<> nathan stratton
On Thu, Oct 15, 2020 at 12:32 PM Ken Rice <krice@freeswitch.org> wrote: Mod_rtmp was intended as a softphone endpoint written in flash. dont use it. litterally load mod_av then via esl, fs_cli etc “conference <confname> [play|record] rtmp://{rest/of/url}”
its really that simple. play will pkay the rtmp stream into the conference like any other media file. record will shoot a media stream out to what ever rtmp reflector you are using auch as youtube or facebook or your own private rtmp server.
Sent from my iPhone
On Oct 15, 2020, at 08:42, Nathan Stratton <nathan@robotics.net> wrote:
On Wed, Oct 14, 2020 at 10:43 PM David P <davidswalkabout@gmail.com> wrote:
In response to the thread " $500 bounty RTMP to conference room", try something like this:
<action application="record_session" data="{modname=mod_av,vw=320,vh=240}rtmp://${ipOfDestination}:${portOfDestination}/${queryPathForDestination}"/>
So I am not trying to send rtmp, I am trying to do I think what mod_rtmp was doing and LISTEN for rtmp that is broadcast to the FreeSWITCH server.
-Nathan _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
You could probably set up ffmpeg as a server to listen for the stream and then pull it on FS. On Thu, Oct 15, 2020 at 9:28 PM Ken Rice <krice@freeswitch.org> wrote:
these is to my knowledge no way to listen for a stream to randomly start up. This would require designing a new module to handle the listening service.
Sent from my iPhone
On Oct 15, 2020, at 11:43, Nathan Stratton <nathan@robotics.net> wrote:
Thanks, so yes, I want to play not record, but when I try to play it's looking for a file, it's not listening on port 1935 for a RTMP stream from the server. I need to ACCEPT a RTMP steam like mod_rtmp did, I understand why mod_rtmp is not working but is there any other way to listen from RTMP on port 1935 and send it into a conference?
freeswitch@troc.staging.vocinity.com> conference 4600 play rtmp://localhost -ERR (play) File: rtmp://localhost not found.
2020-10-15 12:38:12.636530 [ERR] avformat.c:1603 Invalid Format
So the issue is that play rtmp:// does not listen for an incoming RTMP stream.
<> nathan stratton
On Thu, Oct 15, 2020 at 12:32 PM Ken Rice <krice@freeswitch.org> wrote:
Mod_rtmp was intended as a softphone endpoint written in flash. dont use it. litterally load mod_av then via esl, fs_cli etc “conference <confname> [play|record] rtmp://{rest/of/url}”
its really that simple. play will pkay the rtmp stream into the conference like any other media file. record will shoot a media stream out to what ever rtmp reflector you are using auch as youtube or facebook or your own private rtmp server.
Sent from my iPhone
On Oct 15, 2020, at 08:42, Nathan Stratton <nathan@robotics.net> wrote:
On Wed, Oct 14, 2020 at 10:43 PM David P <davidswalkabout@gmail.com> wrote:
In response to the thread " $500 bounty RTMP to conference room", try something like this:
<action application="record_session" data="{modname=mod_av,vw=320,vh=240}rtmp://${ipOfDestination}:${portOfDestination}/${queryPathForDestination}"/>
So I am not trying to send rtmp, I am trying to do I think what mod_rtmp was doing and LISTEN for rtmp that is broadcast to the FreeSWITCH server.
-Nathan _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
I can make ffmpeg listen for RTMP, how would I get that into a conference room?
<> nathan stratton
On Thu, Oct 15, 2020 at 10:36 PM Christopher Rienzo <cmrienzo@gmail.com> wrote:
You could probably set up ffmpeg as a server to listen for the stream and then pull it on FS.
On Thu, Oct 15, 2020 at 9:28 PM Ken Rice <krice@freeswitch.org> wrote:
these is to my knowledge no way to listen for a stream to randomly start up. This would require designing a new module to handle the listening service.
Sent from my iPhone
On Oct 15, 2020, at 11:43, Nathan Stratton <nathan@robotics.net> wrote:
Thanks, so yes, I want to play not record, but when I try to play it's looking for a file, it's not listening on port 1935 for a RTMP stream from the server. I need to ACCEPT a RTMP steam like mod_rtmp did, I understand why mod_rtmp is not working but is there any other way to listen from RTMP on port 1935 and send it into a conference?
freeswitch@troc.staging.vocinity.com> conference 4600 play rtmp://localhost -ERR (play) File: rtmp://localhost not found.
2020-10-15 12:38:12.636530 [ERR] avformat.c:1603 Invalid Format
So the issue is that play rtmp:// does not listen for an incoming RTMP stream.
<> nathan stratton
On Thu, Oct 15, 2020 at 12:32 PM Ken Rice <krice@freeswitch.org> wrote:
Mod_rtmp was intended as a softphone endpoint written in flash. dont use it. litterally load mod_av then via esl, fs_cli etc “conference <confname> [play|record] rtmp://{rest/of/url}”
its really that simple. play will pkay the rtmp stream into the conference like any other media file. record will shoot a media stream out to what ever rtmp reflector you are using auch as youtube or facebook or your own private rtmp server.
Sent from my iPhone
On Oct 15, 2020, at 08:42, Nathan Stratton <nathan@robotics.net> wrote:
On Wed, Oct 14, 2020 at 10:43 PM David P <davidswalkabout@gmail.com> wrote:
In response to the thread " $500 bounty RTMP to conference room", try something like this:
<action application="record_session" data="{modname=mod_av,vw=320,vh=240}rtmp://${ipOfDestination}:${portOfDestination}/${queryPathForDestination}"/>
So I am not trying to send rtmp, I am trying to do I think what mod_rtmp was doing and LISTEN for rtmp that is broadcast to the FreeSWITCH server.
-Nathan _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
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Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
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You think it's worth modifying mod_rtmp? It seems to at least accept a RTMP stream.
<> nathan stratton
On Thu, Oct 15, 2020 at 8:44 PM Ken Rice <krice@freeswitch.org> wrote:
these is to my knowledge no way to listen for a stream to randomly start up. This would require designing a new module to handle the listening service.
Sent from my iPhone
On Oct 15, 2020, at 11:43, Nathan Stratton <nathan@robotics.net> wrote:
Thanks, so yes, I want to play not record, but when I try to play it's looking for a file, it's not listening on port 1935 for a RTMP stream from the server. I need to ACCEPT a RTMP steam like mod_rtmp did, I understand why mod_rtmp is not working but is there any other way to listen from RTMP on port 1935 and send it into a conference?
freeswitch@troc.staging.vocinity.com> conference 4600 play rtmp://localhost -ERR (play) File: rtmp://localhost not found.
2020-10-15 12:38:12.636530 [ERR] avformat.c:1603 Invalid Format
So the issue is that play rtmp:// does not listen for an incoming RTMP stream.
<> nathan stratton
On Thu, Oct 15, 2020 at 12:32 PM Ken Rice <krice@freeswitch.org> wrote:
Mod_rtmp was intended as a softphone endpoint written in flash. dont use it. litterally load mod_av then via esl, fs_cli etc “conference <confname> [play|record] rtmp://{rest/of/url}”
its really that simple. play will pkay the rtmp stream into the conference like any other media file. record will shoot a media stream out to what ever rtmp reflector you are using auch as youtube or facebook or your own private rtmp server.
Sent from my iPhone
On Oct 15, 2020, at 08:42, Nathan Stratton <nathan@robotics.net> wrote:
On Wed, Oct 14, 2020 at 10:43 PM David P <davidswalkabout@gmail.com> wrote:
In response to the thread " $500 bounty RTMP to conference room", try something like this:
<action application="record_session" data="{modname=mod_av,vw=320,vh=240}rtmp://${ipOfDestination}:${portOfDestination}/${queryPathForDestination}"/>
So I am not trying to send rtmp, I am trying to do I think what mod_rtmp was doing and LISTEN for rtmp that is broadcast to the FreeSWITCH server.
-Nathan _________________________________________________________________________
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_________________________________________________________________________
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_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
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_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
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Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
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Hi I can help u Please confirm if bounty is still open On Thu, 15 Oct 2020, 8:59 am David P, <davidswalkabout@gmail.com> wrote:
In response to the thread " $500 bounty RTMP to conference room", try something like this:
<action application="record_session" data="{modname=mod_av,vw=320,vh=240}rtmp://${ipOfDestination}:${portOfDestination}/${queryPathForDestination}"/> _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
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participants (6)
-
Christopher Rienzo
-
David P
-
David Villasmil
-
Ken Rice
-
Nathan Stratton
-
sagar malam