<html><head><meta http-equiv="Content-Type" content="text/html charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;">can you tell us a little more about what exactly is the endpoint that will have "no signaling" it obviously needs some sort of signaling to exchange media information.<div><br><div><div>On Jul 1, 2014, at 8:46 AM, Kees Jan Koster <<a href="mailto:kjkoster@kjkoster.org">kjkoster@kjkoster.org</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div style="font-size: 12px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; white-space: normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;">Dear Stanislav,<br><br>I was afraid you were going to say that. :-/ I guess that's what I will have to do then.<br><br>Kees Jan<br><br><br>On 1 Jul 2014, at 14:40, Stanislav Sinyagin <<a href="mailto:ssinyagin@yahoo.com">ssinyagin@yahoo.com</a>> wrote:<br><br><blockquote type="cite">hmm, then I don't see much other possibilities. Probably it's easier to add the SIP agent functionality to your non-SIP part?<br><br>If it's a Java application, there are multiple implementations of SIP for java.<br><br>From: Kees Jan Koster <<a href="mailto:kjkoster@kjkoster.org">kjkoster@kjkoster.org</a>><br>To: Stanislav Sinyagin <<a href="mailto:ssinyagin@yahoo.com">ssinyagin@yahoo.com</a>><span class="Apple-converted-space"> </span><br>Cc: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><span class="Apple-converted-space"> </span><br>Sent: Tuesday, July 1, 2014 2:34 PM<br>Subject: Re: [Freeswitch-users] handling signalling externally, RTP via freeswitch<br><br>Dear Stanislav,<br><br>No, the audio will be bi-directional. Other than the custom signalling it will be a regular voice call.<br><br>Kees Jan<br><br><br>On 1 Jul 2014, at 14:31, Stanislav Sinyagin <<a href="mailto:ssinyagin@yahoo.com">ssinyagin@yahoo.com</a>> wrote:<br><br><blockquote type="cite">Kees, do I understand it correctly that you're only sending audio towards the SIP endpoint, so it's not a bidirectional conversation?<br><br>In this case, you can utilize audio streaming via HTTP, probably it's the easiest way to deliver audio toward FreeSWITCH:<br><br>session.streamFile("<a href="shout://some.server.com/file.mp3">shout://some.server.com/file.mp3</a>", "")<br><a href="https://wiki.freeswitch.org/wiki/Mod_shout">https://wiki.freeswitch.org/wiki/Mod_shout</a><br><br><br><br><br>From: Kees Jan Koster <kjkoster@java-monitor.com><br>To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>; ssinyagin@yahoo.com<span class="Apple-converted-space"> </span><br>Sent: Tuesday, July 1, 2014 1:27 PM<br>Subject: Re: [Freeswitch-users] handling signalling externally, RTP via freeswitch<br><br>Dear Stanislav,<br><br>I was rather hoping to avoid that path. My C is not all that great. :-/ But if I have to...<br><br>Any endpoints that come close that I might want to look at?<br><br>Kees Jan<br><br><blockquote type="cite">AFAIK the RTP parameters and streaming are handled from the endpoint module:<br>https://wiki.freeswitch.org/wiki/Modules<br><br>so, you might end up in writing your own module for what you want.<br><br><br>From: Kees Jan Koster <kjkoster@kjkoster.org><br>To: freeswitch-users@lists.freeswitch.org<span class="Apple-converted-space"> </span><br>Sent: Tuesday, July 1, 2014 9:21 AM<br>Subject: [Freeswitch-users] handling signalling externally, RTP via freeswitch<br><br>Dear All,<br><br>I need some reading advice/pointers on the following: I want to use FreeSwitch as a bridge from a custom client to SIP. The SIP part is easy and works. The non-SIP part is the challenge.<br><br>I have my audio sending client hooked up via mod_event_socket. I really like the way events and ESL work in FreeSwitch, by the way. Good control and easy to code. But I digress. So I have my client hooked up over ESL for signalling and need to know where to send the RTP stream to from the non-SIP client side.<br><br>Call flow (A-leg is the custom client, B-leg is sofia-SIP)<br>[1] custom client starts call to SIP side over ESL [1]<span class="Apple-converted-space"> </span><br> FreeSwitch handles the signalling with the SIP client<br> SIP client answers the call<br>[2] FreeSwitch sends event which tells the client where the RTP stream should go to<br> custom client starts sending RTP audio<br> FreeSwitch bridges the two legs and people talk<br> either side hangs up and the call is done<br><br>The reverse (SIP dialling the custom client) won't happen and does not have to be supported.<br><br>My concrete question is how [1] should happen. The custom client should issue an "bgapi originate". The arguments for the SIP side are clear, but what to use for the custom client? What should I fill in for [1], below?<br><br>bgapi originate [1] sofia/internal/1002@example.com<br><br>--<br>Kees Jan<br><br>http://java-monitor.com/<br>kjkoster@kjkoster.org<br>+31651838192<br><br>The secret of success lies in the stability of the goal. -- Benjamin Disraeli<br><br><br>_________________________________________________________________________<br>Professional FreeSWITCH Consulting Services:<br>consulting@freeswitch.org<br>http://www.freeswitchsolutions.com<br><br>FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>http://www.cudatel.com<br><br>Official FreeSWITCH Sites<br>http://www.freeswitch.org<br>http://wiki.freeswitch.org<br>http://www.cluecon.com<br><br>FreeSWITCH-users mailing list<br>FreeSWITCH-users@lists.freeswitch.org<br>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>http://www.freeswitch.org<br><br><br>_________________________________________________________________________<br>Professional FreeSWITCH Consulting Services:<br>consulting@freeswitch.org<br>http://www.freeswitchsolutions.com<br><br>FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>http://www.cudatel.com<br><br>Official FreeSWITCH Sites<br>http://www.freeswitch.org<br>http://wiki.freeswitch.org<br>http://www.cluecon.com<br><br>FreeSWITCH-users mailing list<br>FreeSWITCH-users@lists.freeswitch.org<br>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>http://www.freeswitch.org<br></blockquote><br><br>--<br>Kees Jan<br><br>kjkoster@java-monitor.com<br><br>http://java-monitor.com/<br>+31651838192<br><br>The secret of success lies in the stability of the goal. -- Benjamin Disraeli<br></blockquote><br><br><br><blockquote type="cite"><br><br></blockquote><br><br>--<br>Kees Jan<br><br><a href="http://java-monitor.com/">http://java-monitor.com/</a><br>kjkoster@kjkoster.org<br>+31651838192<br><br><br>Change is good. Granted, it is good in retrospect, but change is good.<br><br><br><br></blockquote><br><br>--<br>Kees Jan<br><br><a href="http://java-monitor.com/">http://java-monitor.com/</a><br><a href="mailto:kjkoster@kjkoster.org">kjkoster@kjkoster.org</a><br>+31651838192<br><br>Change is good. Granted, it is good in retrospect, but change is good.<br><br><br>_________________________________________________________________________<br>Professional FreeSWITCH Consulting Services:<br><a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>http://www.freeswitchsolutions.com<br><br>FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>http://www.cudatel.com<br><br>Official FreeSWITCH Sites<br>http://www.freeswitch.org<br>http://wiki.freeswitch.org<br>http://www.cluecon.com<br><br>FreeSWITCH-users mailing list<br>FreeSWITCH-users@lists.freeswitch.org<br>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>http://www.freeswitch.org</div></blockquote></div><br></div></body></html>