In order to really use bypass_media, the two endpoints MUST see each other, no NAT involved.
Although I read somewhere from Brian that parameter does nothing (but I could be misremembering)
In any case, what do you have in your sofia profile in ext-rtp-ip and ext-sip-ip?

On Fri, 20 Nov 2020 at 10:52, Maciej Bylica <mbgatherer@gmail.com> wrote:
Hello,

Sorry for delay, i had to catch up on a few things a bit.
Instead of 79.x.x.x. it should be 10.20.30.41 .... and yes this is what i expect to have.

But i still don't know why FS generates such an event.
As for workaround i use "bypass_media=true" instead of "proxy_media=true", but the question is how to solve it.

Thanks
Maciej



sob., 14 lis 2020 o 09:21 Brian : <brians@iptel.co> napisał(a):
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?

79.x.x.x





On Friday, November 6, 2020, Maciej Bylica <mbgatherer@gmail.com> wrote:
> Hi all,
> I am working on 1.10.5 (today's build) with proxy_media=true configuration set in dialplan config.
> Around 2-3% of total call attempts are CANCELed with "INCOMPATIBLE_DESTINATION" / "AUDIO RTP REPORTS ERROR: [Missing local host]" notification.
> 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.
> I have compared all the SIP call flows incoming and outgoing (INVITES, 183) and found no clue.
>
> Some of the important log snippets:
> - the call is CANCELed once FS receives 183 and gets first early media packets
>
> 8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] switch_ivr_originate.c:3801 Sending early media
> 8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:8729 PROXY AUDIO RTP [sofia/outside_1/4916222112233@10.20.30.20] 10.20.30.6:26502->10.20.30.6:26502 codec: 18 ms: 20
> 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]
> 8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [NOTICE] switch_core_media.c:9670 Hangup sofia/outside_1/4916222112233@10.20.30.20 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
> a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3808 sofia/outside_1/4916222112233@10.20.30.20 Media Establishment Failed.
> 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]
> 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]
> 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
> 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)
> 8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] mod_dptools.c:3631 Originate Failed.  Cause: INCOMPATIBLE_DESTINATION
> a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:12291 sofia/outside_1/3144917111223344 too late.
>
> - incoming (Originator -> FS) SIP INVITE has following SDP
> v=0
> o=- 1604689110 1604689110 IN IP4 10.20.30.6
> s=-
> c=IN IP4 10.20.30.6
> t=0 0
> m=audio 26502 RTP/AVP 18 0 8 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> a=sendrecv
> a=silenceSupp:off - - - -
>
> - outgoing (FS -> Terminator) SDP of SIP INVITE looks like this (codec transparent)
> v=0
> o=FreeSWITCH 4241000132 4241000133 IN IP4 10.20.30.20
> s=FreeSWITCH
> c=IN IP4 10.20.30.20
> t=0 0
> m=audio 31034 RTP/AVP 18 0 8 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> a=silenceSupp:off - - - -
>
> - 183 that is received (Terminator -> FS)
> v=0
> o=JOANO_SDP 24930105 0 IN IP4 10.20.30.40
> s=JOANO-SIP
> c=IN IP4 79.133.196.167
> t=0 0
> m=audio 18200 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=silenceSupp:off - - - -
> a=ptime:20
>
> As a result FS tears down the call attempt by sending (FS -> Originator)
>
> SIP/2.0 488 Not Acceptable Here
> Via: SIP/2.0/UDP 10.20.30.10:5060;received=10.20.30.10;rport=5060;branch=z9hG4bK0609c6993b9c-30-16a290
> Via: SIP/2.0/UDP 10.20.30.2:5060;branch=z9hG4bK5765.136934e2.0
> Max-Forwards: 66
> From: <sip:4916222112233@10.20.30.5:5061;user=phone>;tag=b6b8aec94s
> To: <sip:3144917111223344@10.20.30.20;user=phone>;tag=DcUe4343Uvj8Q
> Call-ID: 684d5998-d859dd12-ca30a685-551b1244
> CSeq: 1 INVITE
> User-Agent: FreeSWITCH-mod_sofia/1.10.5-release~64bit
> Accept: application/sdp
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
> Supported: timer, path, replaces
> Allow-Events: talk, hold, conference, refer
> Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
> Content-Length: 0
>
> and CANCEL (FS -> Terminator)
> CANCEL sip:3144917111223344@10.20.30.30:5060 SIP/2.0
> Via: SIP/2.0/UDP 10.20.30.20:5080;rport;branch=z9hG4bKK01B9mcmpp5jS
> Max-Forwards: 70
> From: <sip:4916222112233@10.20.30.20>;tag=g76r9mQeKQN0a
> To: <sip:3144917111223344@10.20.30.30:5060>
> Call-ID: e788fdfd-9b04-1239-6881-fc15b4264fc0
> CSeq: 27790763 CANCEL
> Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
> Content-Length: 0
>
> Could somebody help me to address the issue I am struggling with ?
>
> Detailed debugging logs and sip signalization might be found here:
> https://pastebin.com/G0sjGiw8 - FS logs
> https://pastebin.com/2Zt6r2mF - SIP logs
>
> Many thanks in advance,
> Maciej
> _________________________________________________________________________

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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

Professional FreeSWITCH Services
sales@freeswitch.com
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

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David Villasmil
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phone: +34669448337