[asterisk-bugs] [JIRA] (ASTERISK-28768) No audio on remote NATted phone when using local_net behind another NATted Asterisk
Asterisk Team (JIRA)
noreply at issues.asterisk.org
Mon Mar 9 04:38:28 CDT 2020
[ https://issues.asterisk.org/jira/browse/ASTERISK-28768?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Asterisk Team updated ASTERISK-28768:
-------------------------------------
Status: Waiting for Feedback (was: Waiting for Feedback)
> No audio on remote NATted phone when using local_net behind another NATted Asterisk
> -----------------------------------------------------------------------------------
>
> Key: ASTERISK-28768
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-28768
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Resources/res_pjsip
> Affects Versions: 16.8.0
> Environment: Asterisk 16.8.0 running on Centos 7 x64.
> Reporter: Marc Ketel
> Assignee: Asterisk Team
> Labels: patch
> Attachments: ASTERISK-27248_undo_undo.patch, Network layout.png, SIP trace asterisk16.8.0 local phone 2995 to asterisk playback - audio works with 2 local_net.txt, SIP trace asterisk16.8.0 local phone 2995 to asterisk playback - audio works with local_net.txt, SIP trace asterisk16.8.0 local phone 2995 to asterisk playback - audio works without local_net.txt, SIP trace asterisk16.8.0 remote phone 2003 to asterisk playback - audio works with 2 local_nets.txt, SIP trace asterisk16.8.0 remote phone 2003 to asterisk playback - audio works without local_net.txt, SIP trace asterisk16.8.0 remote phone 2003 to asterisk playback - no audio with local_net.txt, SIP trace asterisk16.8.0 (with pre-ASTERISK-27248 functionality) local phone 2995 to asterisk playback - audio works with local_net.txt, SIP trace asterisk16.8.0 (with pre-ASTERISK-27248 functionality) remote phone 2003 to asterisk playback - audio works with local_net.txt, SIP trace asterisk16.8.0 (with pre-ASTERISK-27248 functionality) remote phone 2003 to asterisk playback - audio works without local_net.txt
>
>
> The changes in issue https://issues.asterisk.org/jira/browse/ASTERISK-27248 causes local_net to not function correctly.
> In Asterisk 11.10, local_net functioned correctly, in Asterisk 16.8.0 local_net does not function correctly. I traced the change to issue https://issues.asterisk.org/jira/browse/ASTERISK-27248.
> Scenario: both Asterisk and phone1 are behind same NAT. phone2 is remotely behind nat. When configuring some unrelated ip subnet in local_net the remote phone2's audio is being send from Asterisk to the local ip of the remote phone2, which does not work. Disabling local_net makes the remote phone2 work.
> When undoing ASTERISK-27248_undo.patch and configuring local_net to the network of phone1 all phones work. The local phone gets RTP from the local address of Asterisk, the remote phone gets audio to its external nat ip.
> Remote phone with correct external media adress and working RTP
> {noformat}
> [Mar 4 14:16:25] DEBUG[31740]: res_pjsip_session.c:902 handle_negotiated_sdp_session_media: Applied negotiated SDP media stream 'audio' using audio SDP handler
> [Mar 4 14:16:25] DEBUG[31740]: res_pjsip_session.c:3498 handle_outgoing_response: Method is INVITE, Response is 200 OK
> [Mar 4 14:16:25] DEBUG[31740]: res_pjsip/pjsip_message_filter.c:288 filter_on_tx_message: Re-wrote Contact URI host/port to 172.17.32.36:5060 (this may be re-written again later)
> [Mar 4 14:16:25] DEBUG[31740]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting '198.51.100.50' into...
> [Mar 4 14:16:25] DEBUG[31740]: netsock2.c:224 ast_sockaddr_split_hostport: ...host '198.51.100.50' and port ''.
> [Mar 4 14:16:25] DEBUG[31740]: res_pjsip_nat.c:414 process_nat: Re-wrote Contact URI port to 5060
> [Mar 4 14:16:25] DEBUG[31740]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting '172.17.32.36' into...
> [Mar 4 14:16:25] DEBUG[31740]: netsock2.c:224 ast_sockaddr_split_hostport: ...host '172.17.32.36' and port ''.
> [Mar 4 14:16:25] DEBUG[31740]: res_pjsip_session.c:4338 session_outgoing_nat_hook: Setting external media address to 203.0.113.102
> <--- Transmitting SIP response (880 bytes) to TCP:198.51.100.50:56924 --->
> {noformat}
> Local phone that is correctly identified as local with local_net=10.215.152.0/255.255.255.0
> {noformat}
> [Mar 4 14:19:47] DEBUG[31740]: res_pjsip_session.c:902 handle_negotiated_sdp_session_media: Applied negotiated SDP media stream 'audio' using audio SDP handler
> [Mar 4 14:19:47] DEBUG[31740]: res_pjsip_session.c:3498 handle_outgoing_response: Method is INVITE, Response is 200 OK
> [Mar 4 14:19:47] DEBUG[31740]: res_pjsip/pjsip_message_filter.c:288 filter_on_tx_message: Re-wrote Contact URI host/port to 172.17.32.36:5060 (this may be re-written again later)
> [Mar 4 14:19:47] DEBUG[31740]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting '10.215.152.3' into...
> [Mar 4 14:19:47] DEBUG[31740]: netsock2.c:224 ast_sockaddr_split_hostport: ...host '10.215.152.3' and port ''.
> [Mar 4 14:19:47] DEBUG[31740]: res_pjsip_nat.c:403 process_nat: Request is being sent to local address, skipping NAT manipulation
> <--- Transmitting SIP response (797 bytes) to UDP:10.215.152.3:50611 --->
> {noformat}
> {noformat}
> + /* Reversed check here. We don't check the remote
> + * endpoint being in our local net, but whether our
> + * outgoing session IP is local. If it is, we'll do
> + * rewriting. No localnet configured? Always rewrite. */
> {noformat}
> Please reconsider checking the remote ip of the phone and not the outgoing session ip.
> Relevant configuration:
> {noformat}
> [template-transport-ipv6](!)
> ;local_net=
> ;external_media_address=
> ;external_signaling_address=
> [template-transport-ipv4](!)
> local_net=10.215.152.0/255.255.255.0
> external_media_address=203.0.113.102
> external_signaling_address=203.0.113.102
> [template-transport](!)
> type=transport
> external_signaling_port=5060
> allow_reload=yes
> symmetric_transport=yes
> [transport-auto-udp6](template-transport,template-transport-ipv6)
> protocol=udp
> bind=::
> [transport-auto-udp4](template-transport,template-transport-ipv4)
> protocol=udp
> bind=0.0.0.0
> [transport-auto-tcp6](template-transport,template-transport-ipv6)
> protocol=tcp
> bind=::
> [transport-auto-tcp4](template-transport,template-transport-ipv4)
> protocol=tcp
> bind=0.0.0.0
> [user_defaults](!)
> type = wizard
> accepts_registrations = yes
> sends_registrations = no
> accepts_auth = yes
> sends_auth = no
> has_hint = yes
> hint_context = hints
> hint_application = Dial(PJSIP/${EXTEN})
> endpoint/allow = !all,alaw,ulaw
> endpoint/allow_subscribe = yes
> endpoint/allow_transfer = yes
> endpoint/context = uitbellen
> endpoint/device_state_busy_at = 1
> endpoint/direct_media = no
> endpoint/direct_media_method = invite
> endpoint/disable_direct_media_on_nat = yes
> endpoint/force_rport = yes
> endpoint/ice_support = no
> endpoint/inband_progress = yes
> endpoint/moh_suggest = default
> endpoint/rewrite_contact = yes
> endpoint/rtp_ipv6 = yes
> endpoint/rtp_keepalive = 15
> endpoint/rtp_timeout = 60
> endpoint/rtp_timeout_hold = 14400
> endpoint/rtp_symmetric = yes
> endpoint/send_diversion = yes
> endpoint/send_pai = no
> endpoint/send_rpid = no
> endpoint/subscribe_context = hints
> endpoint/trust_id_inbound = no
> endpoint/trust_id_outbound = no
> endpoint/language = nl
> endpoint/send_connected_line = true
> endpoint/trust_connected_line = true
> aor/qualify_frequency = 60
> aor/authenticate_qualify = no
> aor/max_contacts = 1
> aor/remove_existing = yes
> aor/minimum_expiration = 30
> aor/support_path = yes
> ;(Mitel 6865i)
> [2003](user_defaults)
> aor/mailboxes=
> inbound_auth/username=2003
> inbound_auth/password=xxx
> endpoint/dtmf_mode=auto
> endpoint/call_group=
> endpoint/pickup_group=
> ;Marc softphone ()
> [2995](user_defaults)
> aor/mailboxes=
> inbound_auth/username=2995
> inbound_auth/password=xxx
> endpoint/call_group=
> endpoint/pickup_group=
> {noformat}
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