<div><div dir="auto">In order to really use bypass_media, the two endpoints MUST see each other, no NAT involved.</div><div dir="auto">Although I read somewhere from Brian that parameter does nothing (but I could be misremembering)</div><div dir="auto">In any case, what do you have in your sofia profile in ext-rtp-ip and ext-sip-ip?</div></div><div><div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Fri, 20 Nov 2020 at 10:52, Maciej Bylica <<a href="mailto:mbgatherer@gmail.com" target="_blank">mbgatherer@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;padding-left:1ex;border-left-color:rgb(204,204,204)"><div dir="ltr"><div>Hello,</div><div><br></div><div>Sorry for delay, i had to catch up on a few things a bit.</div>Instead of 79.x.x.x. it should be 10.20.30.41 .... and yes this is what i expect to have.<div><br></div><div>But i still don't know why FS generates such an event.</div><div>As for workaround i use "<span style="font-variant-ligatures:no-common-ligatures;font-family:Menlo;font-size:11px;color:rgb(180,36,25)">bypass_media=true</span>" instead of "<span style="font-variant-ligatures:no-common-ligatures;font-family:Menlo;font-size:11px;color:rgb(64,11,217)">proxy_media=true</span>", but the question is how to solve it.</div>
<div><div><br></div><div>Thanks</div></div></div><div dir="ltr"><div><div>Maciej<br><div><div><br></div><div><br></div></div></div></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">sob., 14 lis 2020 o 09:21 Brian : <<a href="mailto:brians@iptel.co" target="_blank">brians@iptel.co</a>> napisał(a):<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;padding-left:1ex;border-left-color:rgb(204,204,204)">I dont know if youre sanitizing ips but the sdp from the 'terminator' has an ip that isnt in any of the other signalling. Is that what you expect? <br><br>79.x.x.x<br><br><br><br><br><br>On Friday, November 6, 2020, Maciej Bylica <<a href="mailto:mbgatherer@gmail.com" target="_blank">mbgatherer@gmail.com</a>> wrote:<br>> Hi all,<br>> I am working on 1.10.5 (today's build) with proxy_media=true configuration set in dialplan config.<br>> Around 2-3% of total call attempts are CANCELed with "INCOMPATIBLE_DESTINATION" / "AUDIO RTP REPORTS ERROR: [Missing local host]" notification.<br>> I assume that the problem is probably closely connected with codec negotiation, but in my case (proxy_media=true) FS does forward the packets onward and endpoints must agree on the same codec to accept the call.<br>> I have compared all the SIP call flows incoming and outgoing (INVITES, 183) and found no clue.<br>><br>> Some of the important log snippets:<br>> - the call is CANCELed once FS receives 183 and gets first early media packets<br>><br>> 8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] switch_ivr_originate.c:3801 Sending early media<br>> 8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:8729 PROXY AUDIO RTP [sofia/outside_1/<a href="mailto:4916222112233@10.20.30.20" target="_blank">4916222112233@10.20.30.20</a>] 10.20.30.6:26502-><a href="http://10.20.30.6:26502" target="_blank">10.20.30.6:26502</a> codec: 18 ms: 20<br>> 8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [ERR] switch_core_media.c:9669 AUDIO RTP REPORTS ERROR: [Missing local host]<br>> 8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [NOTICE] switch_core_media.c:9670 Hangup sofia/outside_1/<a href="mailto:4916222112233@10.20.30.20" target="_blank">4916222112233@10.20.30.20</a> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]<br>> a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3808 sofia/outside_1/<a href="mailto:4916222112233@10.20.30.20" target="_blank">4916222112233@10.20.30.20</a> Media Establishment Failed.<br>> a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [NOTICE] switch_ivr_originate.c:3810 Hangup sofia/outside_1/3144917111223344 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]<br>> 8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3995 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]<br>> a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_channel.c:3565 (sofia/outside_1/3144917111223344) Callstate Change DOWN -> EARLY<br>> a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] sofia.c:7339 sofia/outside_1/3144917111223344 skip receive message [PROGRESS_EVENT] (channel is hungup already)<br>> 8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] mod_dptools.c:3631 Originate Failed. Cause: INCOMPATIBLE_DESTINATION<br>> a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:12291 sofia/outside_1/3144917111223344 too late.<br>><br>> - incoming (Originator -> FS) SIP INVITE has following SDP<br>> v=0<br>> o=- 1604689110 1604689110 IN IP4 10.20.30.6<br>> s=-<br>> c=IN IP4 10.20.30.6<br>> t=0 0<br>> m=audio 26502 RTP/AVP 18 0 8 101<br>> a=rtpmap:18 G729/8000<br>> a=fmtp:18 annexb=no<br>> a=rtpmap:0 PCMU/8000<br>> a=rtpmap:8 PCMA/8000<br>> a=rtpmap:101 telephone-event/8000<br>> a=fmtp:101 0-15<br>> a=ptime:20<br>> a=sendrecv<br>> a=silenceSupp:off - - - -<br>><br>> - outgoing (FS -> Terminator) SDP of SIP INVITE looks like this (codec transparent)<br>> v=0<br>> o=FreeSWITCH 4241000132 4241000133 IN IP4 10.20.30.20<br>> s=FreeSWITCH<br>> c=IN IP4 10.20.30.20<br>> t=0 0<br>> m=audio 31034 RTP/AVP 18 0 8 101<br>> a=rtpmap:18 G729/8000<br>> a=fmtp:18 annexb=no<br>> a=rtpmap:0 PCMU/8000<br>> a=rtpmap:8 PCMA/8000<br>> a=rtpmap:101 telephone-event/8000<br>> a=fmtp:101 0-15<br>> a=ptime:20<br>> a=silenceSupp:off - - - -<br>><br>> - 183 that is received (Terminator -> FS)<br>> v=0<br>> o=JOANO_SDP 24930105 0 IN IP4 10.20.30.40<br>> s=JOANO-SIP<br>> c=IN IP4 79.133.196.167<br>> t=0 0<br>> m=audio 18200 RTP/AVP 8 101<br>> a=rtpmap:8 PCMA/8000<br>> a=rtpmap:101 telephone-event/8000<br>> a=fmtp:101 0-15<br>> a=silenceSupp:off - - - -<br>> a=ptime:20<br>><br>> As a result FS tears down the call attempt by sending (FS -> Originator)<br>><br>> SIP/2.0 488 Not Acceptable Here<br>> Via: SIP/2.0/UDP 10.20.30.10:5060;received=10.20.30.10;rport=5060;branch=z9hG4bK0609c6993b9c-30-16a290<br>> Via: SIP/2.0/UDP 10.20.30.2:5060;branch=z9hG4bK5765.136934e2.0<br>> Max-Forwards: 66<br>> From: <sip:4916222112233@10.20.30.5:5061;user=phone>;tag=b6b8aec94s<br>> To: <<a href="mailto:sip%3A3144917111223344@10.20.30.20" target="_blank">sip:3144917111223344@10.20.30.20</a>;user=phone>;tag=DcUe4343Uvj8Q<br>> Call-ID: 684d5998-d859dd12-ca30a685-551b1244<br>> CSeq: 1 INVITE<br>> User-Agent: FreeSWITCH-mod_sofia/1.10.5-release~64bit<br>> Accept: application/sdp<br>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY<br>> Supported: timer, path, replaces<br>> Allow-Events: talk, hold, conference, refer<br>> Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"<br>> Content-Length: 0<br>><br>> and CANCEL (FS -> Terminator)<br>> CANCEL <a href="http://sip:3144917111223344@10.20.30.30:5060" target="_blank">sip:3144917111223344@10.20.30.30:5060</a> SIP/2.0<br>> Via: SIP/2.0/UDP 10.20.30.20:5080;rport;branch=z9hG4bKK01B9mcmpp5jS<br>> Max-Forwards: 70<br>> From: <<a href="mailto:sip%3A4916222112233@10.20.30.20" target="_blank">sip:4916222112233@10.20.30.20</a>>;tag=g76r9mQeKQN0a<br>> To: <<a href="http://sip:3144917111223344@10.20.30.30:5060" target="_blank">sip:3144917111223344@10.20.30.30:5060</a>><br>> Call-ID: e788fdfd-9b04-1239-6881-fc15b4264fc0<br>> CSeq: 27790763 CANCEL<br>> Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"<br>> Content-Length: 0<br>><br>> Could somebody help me to address the issue I am struggling with ?<br>><br>> Detailed debugging logs and sip signalization might be found here:<br>> <a href="https://pastebin.com/G0sjGiw8" target="_blank">https://pastebin.com/G0sjGiw8</a> - FS logs<br>> <a href="https://pastebin.com/2Zt6r2mF" target="_blank">https://pastebin.com/2Zt6r2mF</a> - SIP logs<br>><br>> Many thanks in advance,<br>> Maciej<br>>
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</div>-- <br><div dir="ltr" class="gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div>Regards,</div><div><br></div>David Villasmil<div>email: <a href="mailto:david.villasmil.work@gmail.com" target="_blank">david.villasmil.work@gmail.com</a></div><div>phone: +34669448337</div></div></div>