<div dir="ltr">I added the <span lang="EN-US">bind_rtp_to_media_address=yes</span> on all endpoints but still the same behaviour. The funny thing is that the G711 audio early media works and doesn't have that Private IP issue. I was also able to cross check with chan_sip on Asterisk 15, exactly the same wrong behaviour. See following capture (PJSIP):<br><div><br>No. Time Source Destination Protocol Length Info<br> 187 2018-04-11 07:19:56.735967 159.89.XX.XX 192.168.1.185 H264 943 PT=H264, SSRC=0x3A7AF929, Seq=27144, Time=1248011648 FU-A<br><br>Frame 187: 943 bytes on wire (7544 bits), 943 bytes captured (7544 bits)<br>Ethernet II, Src: da:81:42:3d:d0:e7 (da:81:42:3d:d0:e7), Dst: IETF-VRRP-VRID_6e (00:00:5e:00:01:6e)<br>Internet Protocol Version 4, Src: 159.89.XX.XX, Dst: 192.168.1.185<br>User Datagram Protocol, Src Port: 11502, Dst Port: 5022<br>Real-Time Transport Protocol<br>H.264<br><br>No. Time Source Destination Protocol Length Info<br> 188 2018-04-11 07:19:56.735993 159.89.XX.XX 192.168.1.185 H264 943 PT=H264, SSRC=0x3A7AF929, Seq=27145, Time=1248011648, Mark FU-A End<br><br>Frame 188: 943 bytes on wire (7544 bits), 943 bytes captured (7544 bits)<br>Ethernet II, Src: da:81:42:3d:d0:e7 (da:81:42:3d:d0:e7), Dst: IETF-VRRP-VRID_6e (00:00:5e:00:01:6e)<br>Internet Protocol Version 4, Src: 159.89.XX.XX, Dst: 192.168.1.185<br>User Datagram Protocol, Src Port: 11502, Dst Port: 5022<br>Real-Time Transport Protocol<br>H.264<br><br>No. Time Source Destination Protocol Length Info<br> 189 2018-04-11 07:19:56.738966 178.82.XX.XX 159.89.XX.XX RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x2A1A1C31, Seq=1820, Time=1104983225<br><br>Frame 189: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits)<br>Ethernet II, Src: JuniperN_34:67:f0 (40:a6:77:34:67:f0), Dst: da:81:42:3d:d0:e7 (da:81:42:3d:d0:e7)<br>Internet Protocol Version 4, Src: 178.82.XX.XX, Dst: 159.89.XX.XX<br>User Datagram Protocol, Src Port: 5020, Dst Port: 16130<br>Real-Time Transport Protocol<br><br>No. Time Source Destination Protocol Length Info<br> 190 2018-04-11 07:19:56.738975 178.82.XX.XX 159.89.XX.XX RTP 214 PT=ITU-T G.722, SSRC=0x49CD55FD, Seq=26679, Time=470333826<br><br>Frame 190: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits)<br>Ethernet II, Src: JuniperN_4f:3f:f0 (40:a6:77:4f:3f:f0), Dst: da:81:42:3d:d0:e7 (da:81:42:3d:d0:e7)<br>Internet Protocol Version 4, Src: 178.82.XX.XX, Dst: 159.89.XX.XX<br>User Datagram Protocol, Src Port: 5004, Dst Port: 18280<br>Real-Time Transport Protocol<br></div></div><div class="gmail_extra"><br><div class="gmail_quote">2018-04-11 9:11 GMT+02:00 Floimair Florian <span dir="ltr"><<a href="mailto:f.floimair@commend.com" target="_blank">f.floimair@commend.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div link="blue" vlink="purple" lang="DE">
<div class="m_3281208218046476239WordSection1">
<p class="MsoNormal"><span lang="EN-US">I did a quick check between what I have set and your settings below.<u></u><u></u></span></p>
<p class="MsoNormal"><span lang="EN-US"><u></u> <u></u></span></p>
<p class="MsoNormal"><span lang="EN-US">You can try the following and see if it helps<u></u><u></u></span></p>
<p class="MsoNormal"><span lang="EN-US"><u></u> <u></u></span></p>
<p class="MsoNormal"><span lang="EN-US">In your endpoint:<br>
bind_rtp_to_media_address=yes<u></u><u></u></span></p><span class="">
<p class="MsoNormal"><span lang="EN-US"><u></u> <u></u></span></p>
<p class="MsoNormal"><span lang="EN-US"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:12.0pt;font-family:"Times New Roman",serif" lang="EN-US"> <u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:12.0pt;font-family:"Times New Roman",serif" lang="EN-US"> <u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial",sans-serif;color:black" lang="EN-US">With best regards<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial",sans-serif;color:black" lang="EN-US"><br>
</span><b><span style="font-size:10.0pt;font-family:"Arial",sans-serif;color:black" lang="EN-US">Florian Floimair<br>
</span></b><span style="font-size:10.0pt;font-family:"Arial",sans-serif;color:black" lang="EN-US">Innovation - Software-Development - VoIP & DevOps</span><span style="font-size:10.0pt;font-family:"Arial",sans-serif;color:black" lang="EN-US"><br>
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</b></span><span style="font-size:8.0pt;font-family:"Arial",sans-serif;color:gray" lang="EN-US">FN 178618z | LG Salzburg</span><span style="font-size:12.0pt;font-family:"Times New Roman",serif" lang="EN-US"><u></u><u></u></span></p>
<p class="MsoNormal"><span lang="EN-US"><u></u> <u></u></span></p>
</span><p class="MsoNormal"><b>Von:</b> <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.<wbr>digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-<wbr>bounces@lists.digium.com</a>]
<b>Im Auftrag von </b>Benjamin Marty<br>
<b>Gesendet:</b> Mittwoch, 11. April 2018 08:55<br>
<b>An:</b> Asterisk Users Mailing List - Non-Commercial Discussion <<a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.<wbr>com</a>></p><div><div class="h5"><br>
<b>Betreff:</b> Re: [asterisk-users] Asterisk behind NAT Early Media Video<u></u><u></u></div></div><p></p><div><div class="h5">
<p class="MsoNormal"><u></u> <u></u></p>
<div>
<div>
<div>
<div>
<div>
<p class="MsoNormal" style="margin-bottom:12.0pt">I think I found the root cause. The H264 Early Media video is received successfully on the Asterisk Server. It also seems to get processed. But it's send to the private IP of the receipent SIP phone.<u></u><u></u></p>
</div>
<p class="MsoNormal">For clarification:<u></u><u></u></p>
</div>
<p class="MsoNormal">178.82.XX.XX is my Public IP of my Internet access. Both phones use this as Public IP via standard Source NAT.<u></u><u></u></p>
</div>
<p class="MsoNormal" style="margin-bottom:12.0pt">159.89.XX.XX is the IP of the Asterisk Server. For this test I used a Server without Destination NAT. So the eth0 interface has this IP.<u></u><u></u></p>
</div>
<p class="MsoNormal">Packet capture:<u></u><u></u></p>
<div>
<div>
<div>
<div>
<p class="MsoNormal" style="margin-bottom:12.0pt">No. Time Source Destination Protocol Length Info<br>
141 2018-04-11 06:40:03.306561 178.82.XX.XX 159.89.XX.XX H264 64 PT=H264, SSRC=0x3CB1E12D, Seq=19561, Time=319121408 SPS<br>
<br>
Frame 141: 64 bytes on wire (512 bits), 64 bytes captured (512 bits)<br>
Ethernet II, Src: JuniperN_34:67:f0 (40:a6:77:34:67:f0), Dst: da:81:42:3d:d0:e7 (da:81:42:3d:d0:e7)<br>
Internet Protocol Version 4, Src: 178.82.169.0, Dst: 159.89.104.193<br>
User Datagram Protocol, Src Port: 5006, Dst Port: 13182<br>
Real-Time Transport Protocol<br>
H.264<br>
<br>
No. Time Source Destination Protocol Length Info<br>
142 2018-04-11 06:40:03.306682 159.89.XX.XX 192.168.XX.XX H264 64 PT=H264, SSRC=0x5EE97C55, Seq=30572, Time=319121408 SPS<br>
<br>
Frame 142: 64 bytes on wire (512 bits), 64 bytes captured (512 bits)<br>
Ethernet II, Src: da:81:42:3d:d0:e7 (da:81:42:3d:d0:e7), Dst: IETF-VRRP-VRID_6e (00:00:5e:00:01:6e)<br>
Internet Protocol Version 4, Src: 159.89.104.193, Dst: 192.168.1.185<br>
User Datagram Protocol, Src Port: 10298, Dst Port: 5022<br>
Real-Time Transport Protocol<br>
H.264<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal" style="margin-bottom:12.0pt">PJSIP.conf:<br>
[7004]<br>
type = endpoint<br>
context = internal<br>
rewrite_contact = yes<br>
direct_media = no<br>
rtp_symmetric = yes<br>
;force_rport = yes<br>
disallow = all<br>
allow = g722, alaw, ulaw, gsm, ilbc, h264<br>
aors = 7004<br>
auth = auth7004<br>
<br>
[7004]<br>
type = aor<br>
max_contacts = 2<br>
<br>
[auth7004]<br>
type=auth<br>
auth_type=userpass<br>
password=1234<br>
username=7004<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal">extensions.conf:<br>
[internal]<br>
exten => _700X,1,Dial(PJSIP/${EXTEN})<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal" style="margin-bottom:12.0pt"><u></u> <u></u></p>
</div>
</div>
</div>
</div>
</div>
<div>
<p class="MsoNormal"><u></u> <u></u></p>
<div>
<p class="MsoNormal">2018-04-10 16:43 GMT+02:00 Benjamin Marty <<a href="mailto:benjamin.marty@gmail.com" target="_blank">benjamin.marty@gmail.com</a>>:<u></u><u></u></p>
<blockquote style="border:none;border-left:solid #cccccc 1.0pt;padding:0cm 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<p class="MsoNormal" style="margin-bottom:12.0pt">I just noticed, the calling device isn't even sending the early media video stream. It just sends an early media audio stream. Is there propably a change in the signaling needed?<u></u><u></u></p>
</div>
<p class="MsoNormal">(On another P2P SIP Server the early media video works.)<u></u><u></u></p>
</div>
<div>
<div>
<div>
<p class="MsoNormal"><u></u> <u></u></p>
<div>
<p class="MsoNormal">2018-04-10 12:29 GMT+02:00 Benjamin Marty <<a href="mailto:benjamin.marty@gmail.com" target="_blank">benjamin.marty@gmail.com</a>>:<u></u><u></u></p>
<blockquote style="border:none;border-left:solid #cccccc 1.0pt;padding:0cm 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<div>
<div>
<div>
<p class="MsoNormal" style="margin-bottom:12.0pt">Hi Florian<u></u><u></u></p>
</div>
<p class="MsoNormal" style="margin-bottom:12.0pt">I already have the external_media_address set in the PJSIP setup. Also the external_signaling_address is set to the Public IP. If I make a call from an Early Media (video&audio) capable device to an Early Media
capable device (also video&audio) the Early Media audio works perfectly. But no video. If I sniff with wireshark on the recipent device I just see G711 (audio) RTP traffic. The h264 RTP traffic is missing before I accept the call. After accepting the call
the h264 RTP traffic comes through.<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal">The 183 SIP protocoll comes through. Even Asterisk is noticing it:<br>
-- PJSIP/6002-00000013 is making progress passing it to PJSIP/6001-00000012<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal"><u></u> <u></u></p>
</div>
<p class="MsoNormal" style="margin-bottom:12.0pt">I tried both Asterisk 15 with pjsip.conf configuration and Asterisk 13 with sip.conf (chan_sip). In both cases I just put the both case AST_FRAME_VIDEO: statements before the two voice cases, like in your diff
and recompiled/reinstalled.<u></u><u></u></p>
</div>
<p class="MsoNormal" style="margin-bottom:12.0pt">Regards<u></u><u></u></p>
</div>
<p class="MsoNormal"><span class="m_3281208218046476239m-8827279076902006552hoenzb"><span style="color:#888888">Benjamin<u></u><u></u></span></span></p>
<div>
<div>
<p class="MsoNormal" style="margin-bottom:12.0pt"><u></u> <u></u></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class="MsoNormal"><u></u> <u></u></p>
<div>
<p class="MsoNormal">2018-04-10 9:37 GMT+02:00 Floimair Florian <<a href="mailto:f.floimair@commend.com" target="_blank">f.floimair@commend.com</a>>:<u></u><u></u></p>
<blockquote style="border:none;border-left:solid #cccccc 1.0pt;padding:0cm 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<p class="MsoNormal">Hi Benjamin!<br>
<br>
You're obviously using a similar scenario that I have in place for testing.<br>
I initially had issues with early media (not only video also audio) as well in that scenario. What I had to do was to additionally set<br>
<br>
external_media_address=<your external IP><br>
<br>
in pjsip.conf<br>
<br>
Also, as I wrote the patch for early-media video I'd be interested in any feedback from it.<br>
<br>
<br>
<br>
<br>
With best regards<br>
<br>
Florian Floimair<br>
Innovation - Software-Development - VoIP & DevOps<br>
<br>
COMMEND INTERNATIONAL GMBH<br>
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-----Ursprüngliche Nachricht-----<br>
Von: <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.<wbr>digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-<wbr>bounces@lists.digium.com</a>] Im Auftrag
von Joshua Colp<br>
Gesendet: Montag, 9. April 2018 18:15<br>
An: <a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.<wbr>com</a><br>
Betreff: Re: [asterisk-users] Asterisk behind NAT Early Media Video<br>
<br>
On Mon, Apr 9, 2018, at 1:05 PM, Benjamin Marty wrote:<br>
> wohoo, so if I unterstand it correctly with that patch early media<br>
> video works over the Asterisk server? In other words the Asterisk<br>
> server get's able to (process/)forward the early media video stream with that patch?<br>
<br>
The patch forwards video while in an early media state before the call is answered and bridged, yes.<br>
<br>
--<br>
Joshua Colp<br>
Digium, Inc. | Senior Software Developer<br>
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<p class="MsoNormal"><u></u> <u></u></p>
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