[Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO RTP REPORTS ERROR [Missing local host]

Fri Nov 20 11:09:47 UTC 2020

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 at 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 at 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 at 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 at 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 at 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 at 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 at 10.20.30.5:5061;user=phone>;tag=b6b8aec94s
>> > To: <sip:3144917111223344 at 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 at 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 at 10.20.30.20>;tag=g76r9mQeKQN0a
>> > To: <sip:3144917111223344 at 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 at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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