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

Maciej Bylica mbgatherer at gmail.com
Fri Nov 6 20:46:04 UTC 2020


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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