<html><head><meta http-equiv="Content-Type" content="text/html charset=windows-1252"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">Yes, I already said the most trivial way to do it. &nbsp;Send webrtc traffic and registers to freeswitch directly on websockets, or if not, dynamically generate dialplan to send invites to webrtc hosts with that variable.<div class=""><br class=""><div class=""><br class=""><div><blockquote type="cite" class=""><div class="">On May 31, 2016, at 1:04 PM, Jn Fri &lt;<a href="mailto:furi@vmtele.com" class="">furi@vmtele.com</a>&gt; wrote:</div><br class="Apple-interchange-newline"><div class="">
  
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    I asked anybody whether there is a way how to achieve my goal.
    Somebody that is more skilled than I am.<br class="">
    I was asking if there are "trivially easy ways to accomplish my
    goal". Freeswitch is an awesome software, I like it very much, but
    replacing kamailio with freeswitch is not a solution for me (in our
    environment).<br class="">
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    <br class="">
    On 31.05.2016 18:28, Michael Jerris wrote:<br class="">
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      Its all software, anything is possible with enough code
      modification. &nbsp;We don't have a setting to do those things right
      now, as it doesn't make any sense when there are trivially easy
      ways to accomplish the same goal without writing any more code.
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            <div class="">On May 31, 2016, at 12:07 PM, Jn Fri &lt;<a moz-do-not-send="true" href="mailto:furi@vmtele.com" class=""></a><a class="moz-txt-link-abbreviated" href="mailto:furi@vmtele.com">furi@vmtele.com</a>&gt; wrote:</div>
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              <div text="#000000" bgcolor="#FFFFFF" class=""> Thank you
                Michael, <br class="">
                I understand, so isn't it possible to have SDP with
                m=audio lines SAVPF, SAVP and AVP all together ?<br class="">
                Because I set media_webrtc=true, AVP and SAVP lines are
                replaced with SAVPF.<br class="">
                If I had all three m=audio proto lines I was able to
                manage it with Kamailio.<br class="">
                <br class="">
                And back to my first question, is that possible to set
                different ports&nbsp; for m=audio lines in SDP ?<br class="">
                <br class="">
                Jan<br class="">
                <br class="">
                <div class="moz-cite-prefix">On 31.05.2016 17:16,
                  Michael Jerris wrote:<br class="">
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                  If you don't use some other engine to handle webrtc,
                  then if you are calling something that is registered
                  to you over websockets it will automatically enable
                  media_webrtc for you. &nbsp;Otherwise you'll need some
                  external way of knowing if the endpoint is webrtc or
                  not so you can apply the settings properly.
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                        <div class="">On May 30, 2016, at 9:14 AM, Jn
                          Fri &lt;<a moz-do-not-send="true" class="moz-txt-link-abbreviated" href="mailto:furi@vmtele.com">furi@vmtele.com</a>&gt;
                          wrote:</div>
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                            Hello, <br class="">
                            <br class="">
                            My freeswitch's INVITE sdp offers two
                            m=audio lines. (RTP/SAVP and RTP/AVP). <br class="">
                            That's OK, because I want it so ;)<br class="">
                            <br class="">
                            But both use the same audio port number. <br class="">
                            Is that possible to change this behavior ?
                            To use different audio port numbers for each
                            m=audio line ...<br class="">
                            Please check my example below. In SDP from
                            my freeswitch, both m-lines have <b class="">m=audio </b><b class="">22284</b>.<br class="">
                            <br class="">
                            The reason, why I want to change this is
                            WebRTC and RTPengine. RTPengine changes this
                            INVITEs to RTP/SAVPF.<br class="">
                            And the changed RTP/SAVPF is then not
                            acceptable for chrome browsers. Firefox
                            works well (so far).<br class="">
                            I already reported this to the RTPengine but
                            they say the problem is with INVITE from
                            freeswitch because the m-audio lines use the
                            same port.<br class="">
                            <br class="">
                            My example :<br class="">
                            incoming calls -&gt; Media Server
                            (freeswitch) -&gt; Kamailio (sip and
                            websocket proxy) -&gt; WebRTP and SIP
                            clients <br class="">
                            <br class="">
                            I have found a feature <b class="">media_webrtc=true</b>
                            so I could avoid using rtpengine for webrtc
                            clients (that would be really awesome), but
                            then classic sip clients are offered only
                            with RTP/SAVPF and most sip phones do not
                            know RTP/SAVPF so the calls are rejected :(<br class="">
                            <br class="">
                            <br class="">
                            INVITE SDP from freeswitch: <br class="">
                            <br class="">
                            v=0.<br class="">
                            o=SBC 1464589775 1464589776 IN IP4
                            &lt;freeswitch-public-ip&gt;.<br class="">
                            s=SBC.<br class="">
                            c=IN IP4 &lt;freeswitch-public-ip&gt;.<br class="">
                            t=0 0.<br class="">
                            <b class="">m=audio 22284 RTP/SAVP 8 0 9 3
                              18.</b><br class="">
                            a=rtpmap:8 PCMA/8000.<br class="">
                            a=rtpmap:0 PCMU/8000.<br class="">
                            a=rtpmap:9 G722/8000.<br class="">
                            a=rtpmap:3 GSM/8000.<br class="">
                            a=rtpmap:18 G729/8000.<br class="">
                            a=rtcp:22285 IN IP4
                            &lt;freeswitch-public-ip&gt;.<br class="">
                            a=crypto:1 AEAD_AES_256_GCM_8
                            inline:Q/O+iUzZ9AZLcrADA7w/XcpkmuW6yArr1vsaWc0coTyWUIRLX2qfCA7XDAs.<br class="">
                            a=crypto:2 AEAD_AES_128_GCM_8
                            inline:mBb/SmBuSQNSm8lC5giUfZMCv0ZYINvsQiX2sw.<br class="">
                            a=crypto:3 AES_CM_256_HMAC_SHA1_80
                            inline:ahOTNSsdmHLIOBqvUyGyNjd+gDfAE/+jA6w7XwzGWxBMjtzf5akXNvM/OGy0jQ.<br class="">
                            a=crypto:4 AES_CM_192_HMAC_SHA1_80
                            inline:k7002pXV/SUT7JHhZTYaMV5keUTp1EP57M4rcNZnAoZZzsceTXA.<br class="">
                            a=crypto:5 AES_CM_128_HMAC_SHA1_80
                            inline:bHcV4E/OnzMkNeaFplPWt4RELILYZeGlifnJNlRV.<br class="">
                            a=crypto:6 AES_CM_256_HMAC_SHA1_32
                            inline:zJkowU1tc5rQR5BPpg2m3eE97ZqXLFFJc1Agh89XuZynPFMrXhO266+eMZCd2A.<br class="">
                            a=crypto:7 AES_CM_192_HMAC_SHA1_32
                            inline:2DwGI3UAPsLFBS5sJBsc+pzEsITQwDHCvB0u7pK/XEE3G8swsrw.<br class="">
                            a=crypto:8 AES_CM_128_HMAC_SHA1_32
                            inline:Jen6Gm56Z+4RTRmfpXTpSpoG5bduyythl1j21a15.<br class="">
                            a=crypto:9 AES_CM_128_NULL_AUTH
                            inline:Nn83liGQY7eY8LGs9qz/7EoPiHOpRgwY8H7Ts/b5.<br class="">
                            a=ptime:20.<br class="">
                            <b class="">m=audio 22284 RTP/AVP 8 0 9 3
                              18.</b><br class="">
                            a=rtpmap:8 PCMA/8000.<br class="">
                            a=rtpmap:0 PCMU/8000.<br class="">
                            a=rtpmap:9 G722/8000.<br class="">
                            a=rtpmap:3 GSM/8000.<br class="">
                            a=rtpmap:18 G729/8000.<br class="">
                            a=rtcp:22285 IN IP4
                            &lt;freeswitch-public-ip&gt;.<br class="">
                            a=ptime:20.<br class="">
                            <br class="">
                            <br class="">
                            <br class="">
                            <br class="">
                            INVITE SDP from RTPengine :<br class="">
                            <br class="">
                            v=0<br class="">
                            o=SBC 1464589775 1464589776 IN IP4
                            212.232.17.66<br class="">
                            s=SBC<br class="">
                            c=IN IP4 &lt;rtpengine-public-ip&gt;<br class="">
                            t=0 0<br class="">
                            <b class="">m=audio 31836 RTP/SAVPF 8 0 9 3
                              18</b><br class="">
                            a=rtpmap:8 PCMA/8000<br class="">
                            a=rtpmap:0 PCMU/8000<br class="">
                            a=rtpmap:9 G722/8000<br class="">
                            a=rtpmap:3 GSM/8000<br class="">
                            a=rtpmap:18 G729/8000<br class="">
                            a=ptime:20<br class="">
                            a=sendrecv<br class="">
                            a=rtcp:31837<br class="">
                            a=setup:actpass<br class="">
                            a=fingerprint:sha-1
                            11:76:2D:2A:F7:0D:5A:23:9D:F6:0C:E7:4C:DF:1E:CB:BF:5D:76:4F<br class="">
                            a=ice-ufrag:NauDcjVU<br class="">
                            a=ice-pwd:AiHk6LF4tE0GElyFMWtLeod1wH<br class="">
                            a=candidate:cE0FGbXWIfwj6OGD 1 UDP
                            2130706431 &lt;rtpengine-public-ip&gt; 31836
                            typ host<br class="">
                            a=candidate:cE0FGbXWIfwj6OGD 2 UDP
                            2130706430 &lt;rtpengine-public-ip&gt; 31837
                            typ host<br class="">
                            <b class="">m=audio 31866 RTP/SAVPF 8 0 9 3
                              18</b><br class="">
                            a=rtpmap:8 PCMA/8000<br class="">
                            a=rtpmap:0 PCMU/8000<br class="">
                            a=rtpmap:9 G722/8000<br class="">
                            a=rtpmap:3 GSM/8000<br class="">
                            a=rtpmap:18 G729/8000<br class="">
                            a=ptime:20<br class="">
                            a=sendrecv<br class="">
                            a=rtcp:31867<br class="">
                            a=setup:actpass<br class="">
                            a=fingerprint:sha-1
                            11:76:2D:2A:F7:0D:5A:23:9D:F6:0C:E7:4C:DF:1E:CB:BF:5D:76:4F<br class="">
                            a=ice-ufrag:YxCyPhQV<br class="">
                            a=ice-pwd:2vYt8YaMhIp2DSLPeYOKDqqBX0<br class="">
                            a=candidate:cE0FGbXWIfwj6OGD 1 UDP
                            2130706431 &lt;rtpengine-public-ip&gt; 31866
                            typ host<br class="">
                            a=candidate:cE0FGbXWIfwj6OGD 2 UDP
                            2130706430 &lt;rtpengine-public-ip&gt; 31867
                            typ host<br class="">
                            <br class="">
                          </div>
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