Hi,
I have a problem in changing Port 5060 to a new port like 4849, after changing the port the connection works in the local network, but when connecting from the internet, there is no voice only connects without voice
I hope for help
Note i have Firewall Mikrotik .
Br : Abdulkader Alrazj
Hello !
Your port 5060 or the newer 4849 is just related to SIP messages… You can start testing if your extensions can call each other… answer calls… If these actions are working fine it means that your port 4849 is good. Now you are good to check the RTP side, that cares about voice. First try to check your /asterisk/rtp.conf there find rtpstart and rtpend these parameters show you your RTP range that must be configured in your NAT… If you check and sounds to be good, try to capture tcpdump and check SDP negotiation port… It is a start …
/asterisk/rtp.conf
rtpstart=10000
rtpend=20000
how to configure rtp in NAT !?
NAT is configured in your Firewall… Don’t forget to use in your sip.conf
[general]
Externip= 123.123.123.123;external ip on your NAT device
Localnet= 192.168.0.0/255.255.255.0; local asterisk network
Nat=yes.
When i use port 5060 in sip.conf, the sound works but when you change port 5060 to 4849, the connection works but without sound. Note When i connect from the LAN, the connection works without sound, but when i connect from the Internet, there is no connection & voice
asterisk 13
Try to capture tcpdump and asterisk logs from your call and post it.
== Using SIP RTP CoS mark 5
– Executing [173@DLPN_DialPlan1:1] Dial(“SIP/170-0000006c”, “SIP/173”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/173
– SIP/173-0000006d is ringing
– SIP/173-0000006d is ringing
– SIP/173-0000006d answered SIP/170-0000006c
– Channel SIP/173-0000006d joined ‘simple_bridge’ basic-bridge <36ae74ef-e943-42a5-955c-71d22582acf1>
– Channel SIP/170-0000006c joined ‘simple_bridge’ basic-bridge <36ae74ef-e943-42a5-955c-71d22582acf1>
Note : 173 = Net 170 = Local
When use the default port 5060, the connection and the voice work from both ends Without any problem
My guess is that your router firewall/NAT function is reading SIP traffic and opening up ports/setting up translations for the media streams.
When you use a non-standard port, the router treats it as an unknown type of UDP traffice.
These settings are deprecated on Asterisk 13.x and >
I did not get the correct answer is still the problem
i have Mikrotik For NAT
When i use port 5060 in sip.conf, the sound works but when you change port 5060 to 4849, the connection works but without sound. Note When i connect from the LAN, the connection works without sound, but when i connect from the Internet, there is no connection & voice
i have mikrotik firewall for NAT + asterisk 13
Local IP for asterisk : 192.168.3.15
Router DSL > 192.168.221.60 DMZ To Mikroitk
Mikrotik NAT = dstnat UDP 5060 dst-nat 192.168.3.15 5060
Please provide:
The complete SDP exchange for a failed call (sip set debug on);
Details of the rules on the router for both RTP and port 4849.
Evidence to show what port numbers are actually being used for RTP in both directions.
A rationale for why the router would not need to create dynamic rules for RTP might also help.
*CLI> sip set debug on
SIP Debugging enabled
<— SIP read from UDP:192.168.3.1:42930 —>
<------------->
Really destroying SIP dialog ‘6a0b3d66-507a-4622-b09a-508d1b88d67e’ Method: REGISTER
<— SIP read from UDP:192.168.3.1:4849 —>
INVITE sip:173@192.168.3.15:4849 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.19:4849;rport;branch=z9hG4bK1296359165
From: “170” sip:170@192.168.3.15:4849;tag=97364269
To: sip:173@192.168.3.15:4849
Call-ID: 1999131857@192.168.4.19
CSeq: 1 INVITE
Contact: sip:170@192.168.4.19:4849
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink IP116 2.60.4.5
Supported: replaces
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 234
v=0
o=- 20067 20067 IN IP4 192.168.4.19
s=SDP data
c=IN IP4 192.168.4.19
t=0 0
m=audio 11780 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
<------------->
— (14 headers 12 lines) —
Sending to 192.168.3.1:4849 (NAT)
Sending to 192.168.3.1:4849 (NAT)
Using INVITE request as basis request - 1999131857@192.168.4.19
Found peer ‘170’ for ‘170’ from 192.168.3.1:4849
== Using SIP RTP CoS mark 5
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (g729|ulaw), peer - audio=(g729)/video=(nothing)/text=(nothing), combined - (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> 0xb302b368 – Strict RTP learning after remote address set to: 192.168.4.19:11780
Peer audio RTP is at port 192.168.4.19:11780
Looking for 173 in DLPN_DialPlan1 (domain 192.168.3.15)
sip_route_dump: route/path hop: sip:170@192.168.4.19:4849
<— Transmitting (NAT) to 192.168.3.1:4849 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.4.19:4849;branch=z9hG4bK1296359165;received=192.168.3.1;rport=4849
From: “170” sip:170@192.168.3.15:4849;tag=97364269
To: sip:173@192.168.3.15:4849
Call-ID: 1999131857@192.168.4.19
CSeq: 1 INVITE
Server: Asterisk PBX 13.22.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:173@192.168.3.15:4849
Content-Length: 0
<------------>
– Executing [173@DLPN_DialPlan1:1] Dial(“SIP/170-0000000f”, “SIP/173”) in new stack
== Using SIP RTP CoS mark 5
Audio is at 12448
Adding codec g729 to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.3.1:42930:
INVITE sip:173@192.168.3.1:42930;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.3.15:4849;branch=z9hG4bK4669e74b;rport
Max-Forwards: 70
From: “170” sip:170@192.168.3.15:4849;tag=as774622c9
To: sip:173@192.168.3.1:42930;ob
Contact: sip:170@192.168.3.15:4849
Call-ID: 5b26a95f425f0093306794272420cc87@192.168.3.15:4849
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.22.0
Date: Thu, 02 Aug 2018 09:50:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: “170” sip:170@192.168.3.15;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 287
v=0
o=root 1829924761 1829924761 IN IP4 192.168.3.15
s=Asterisk PBX 13.22.0
c=IN IP4 192.168.3.15
t=0 0
m=audio 12448 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
-- Called SIP/173
Really destroying SIP dialog ‘f14062a5-eca0-4773-a3a8-25aa5c5f122f’ Method: REGISTER
<— SIP read from UDP:192.168.3.1:42930 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.3.15:4849;rport=4849;received=212.154.77.116;branch=z9hG4bK4669e74b
Call-ID: 5b26a95f425f0093306794272420cc87@192.168.3.15:4849
From: “170” sip:170@192.168.221.60;tag=as774622c9
To: sip:173@5.24.31.55;ob
CSeq: 102 INVITE
Content-Length: 0
<------------->
— (7 headers 0 lines) —
<— SIP read from UDP:192.168.3.1:42930 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.3.15:4849;rport=4849;received=212.154.77.116;branch=z9hG4bK4669e74b
Call-ID: 5b26a95f425f0093306794272420cc87@192.168.3.15:4849
From: “170” sip:170@192.168.221.60;tag=as774622c9
To: sip:173@5.24.31.55;ob;tag=32dccd75-95d7-4a8e-b972-8a9a386b245c
CSeq: 102 INVITE
Contact: sip:173@192.168.3.1:42930;ob
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0
<------------->
— (9 headers 0 lines) —
sip_route_dump: route/path hop: sip:173@192.168.3.1:42930;ob
– SIP/173-00000010 is ringing
<— Transmitting (NAT) to 192.168.3.1:4849 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.4.19:4849;branch=z9hG4bK1296359165;received=192.168.3.1;rport=4849
From: “170” sip:170@192.168.3.15:4849;tag=97364269
To: sip:173@192.168.3.15:4849;tag=as343f6e44
Call-ID: 1999131857@192.168.4.19
CSeq: 1 INVITE
Server: Asterisk PBX 13.22.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:173@192.168.3.15:4849
Remote-Party-ID: “173” sip:173@192.168.3.15;party=called;privacy=off;screen=no
Content-Length: 0
<------------>
<— SIP read from UDP:192.168.3.1:42930 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.15:4849;rport=4849;received=212.154.77.116;branch=z9hG4bK4669e74b
Call-ID: 5b26a95f425f0093306794272420cc87@192.168.3.15:4849
From: “170” sip:170@192.168.221.60;tag=as774622c9
To: sip:173@5.24.31.55;ob;tag=32dccd75-95d7-4a8e-b972-8a9a386b245c
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: sip:173@192.168.3.1:42930;ob
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 303
v=0
o=- 3742192220 3742192221 IN IP4 10.191.220.191
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4002 RTP/AVP 18 101
c=IN IP4 10.191.220.191
b=TIAS:64000
a=rtcp:4003 IN IP4 10.191.220.191
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
— (11 headers 15 lines) —
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (g729|ulaw), peer - audio=(g729)/video=(nothing)/text=(nothing), combined - (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> 0xb2e1ee28 – Strict RTP learning after remote address set to: 10.191.220.191:4002
Peer audio RTP is at port 10.191.220.191:4002
sip_route_dump: route/path hop: sip:173@192.168.3.1:42930;ob
Transmitting (NAT) to 192.168.3.1:42930:
ACK sip:173@192.168.3.1:42930;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.3.15:4849;branch=z9hG4bK7747e197;rport
Max-Forwards: 70
From: “170” sip:170@192.168.3.15:4849;tag=as774622c9
To: sip:173@192.168.3.1:42930;ob;tag=32dccd75-95d7-4a8e-b972-8a9a386b245c
Contact: sip:170@192.168.3.15:4849
Call-ID: 5b26a95f425f0093306794272420cc87@192.168.3.15:4849
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.22.0
Content-Length: 0
-- SIP/173-00000010 answered SIP/170-0000000f
Audio is at 18672
Adding codec g729 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Reliably Transmitting (NAT) to 192.168.3.1:4849 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.4.19:4849;branch=z9hG4bK1296359165;received=192.168.3.1;rport=4849
From: “170” sip:170@192.168.3.15:4849;tag=97364269
To: sip:173@192.168.3.15:4849;tag=as343f6e44
Call-ID: 1999131857@192.168.4.19
CSeq: 1 INVITE
Server: Asterisk PBX 13.22.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:173@192.168.3.15:4849
Remote-Party-ID: “173” sip:173@192.168.3.15;party=called;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 263
v=0
o=root 2054220922 2054220922 IN IP4 192.168.3.15
s=Asterisk PBX 13.22.0
c=IN IP4 192.168.3.15
t=0 0
m=audio 18672 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:230
a=sendrecv
<------------>
– Channel SIP/173-00000010 joined ‘simple_bridge’ basic-bridge <738a7ee2-cabe-49ed-be7e-96699d203912>
– Channel SIP/170-0000000f joined ‘simple_bridge’ basic-bridge <738a7ee2-cabe-49ed-be7e-96699d203912>
> Bridge 738a7ee2-cabe-49ed-be7e-96699d203912: switching from simple_bridge technology to native_rtp
> Locally RTP bridged ‘SIP/170-0000000f’ and ‘SIP/173-00000010’ in stack
<— SIP read from UDP:192.168.3.1:4849 —>
ACK sip:173@192.168.3.15:4849 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.19:4849;rport;branch=z9hG4bK1692180488
From: “170” sip:170@192.168.3.15:4849;tag=97364269
To: sip:173@192.168.3.15:4849;tag=as343f6e44
Call-ID: 1999131857@192.168.4.19
CSeq: 1 ACK
Contact: sip:170@192.168.4.19:4849
Max-Forwards: 70
User-Agent: Yealink IP116 2.60.4.5
Content-Length: 0
<------------->
— (10 headers 0 lines) —
<— SIP read from UDP:192.168.3.1:4849 —>
BYE sip:173@192.168.3.15:4849 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.19:4849;rport;branch=z9hG4bK881813300
From: “170” sip:170@192.168.3.15:4849;tag=97364269
To: sip:173@192.168.3.15:4849;tag=as343f6e44
Call-ID: 1999131857@192.168.4.19
CSeq: 2 BYE
Contact: sip:170@192.168.4.19:4849
Max-Forwards: 70
User-Agent: Yealink IP116 2.60.4.5
Content-Length: 0
<------------->
— (10 headers 0 lines) —
Sending to 192.168.3.1:4849 (NAT)
Scheduling destruction of SIP dialog ‘1999131857@192.168.4.19’ in 32000 ms (Method: BYE)
<— Transmitting (NAT) to 192.168.3.1:4849 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.4.19:4849;branch=z9hG4bK881813300;received=192.168.3.1;rport=4849
From: “170” sip:170@192.168.3.15:4849;tag=97364269
To: sip:173@192.168.3.15:4849;tag=as343f6e44
Call-ID: 1999131857@192.168.4.19
CSeq: 2 BYE
Server: Asterisk PBX 13.22.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
– Channel SIP/170-0000000f left ‘native_rtp’ basic-bridge <738a7ee2-cabe-49ed-be7e-96699d203912>
– Channel SIP/173-00000010 left ‘native_rtp’ basic-bridge <738a7ee2-cabe-49ed-be7e-96699d203912>
== Spawn extension (DLPN_DialPlan1, 173, 1) exited non-zero on ‘SIP/170-0000000f’
Scheduling destruction of SIP dialog ‘5b26a95f425f0093306794272420cc87@192.168.3.15:4849’ in 32000 ms (Method: INVITE)
Reliably Transmitting (NAT) to 192.168.3.1:42930:
BYE sip:173@192.168.3.1:42930;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.3.15:4849;branch=z9hG4bK3fb60d99;rport
Max-Forwards: 70
From: “170” sip:170@192.168.3.15:4849;tag=as774622c9
To: sip:173@192.168.3.1:42930;ob;tag=32dccd75-95d7-4a8e-b972-8a9a386b245c
Call-ID: 5b26a95f425f0093306794272420cc87@192.168.3.15:4849
CSeq: 103 BYE
User-Agent: Asterisk PBX 13.22.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
<— SIP read from UDP:192.168.3.1:42930 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.15:4849;rport=4849;received=212.154.77.116;branch=z9hG4bK3fb60d99
Call-ID: 5b26a95f425f0093306794272420cc87@192.168.3.15:4849
From: “170” sip:170@192.168.221.60;tag=as774622c9
To: sip:173@5.24.31.55;ob;tag=32dccd75-95d7-4a8e-b972-8a9a386b245c
CSeq: 103 BYE
Content-Length: 0
<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘5b26a95f425f0093306794272420cc87@192.168.3.15:4849’ Method: INVITE
<— SIP read from UDP:192.168.3.1:42930 —>
<------------->
<— SIP read from UDP:192.168.3.1:4849 —>
<------------->
Really destroying SIP dialog ‘2085179815@192.168.4.19’ Method: REGISTER
<— SIP read from UDP:192.168.3.1:42930 —>
<------------->
Really destroying SIP dialog ‘1999131857@192.168.4.19’ Method: BYE
<— SIP read from UDP:192.168.3.1:42930 —>
<------------->
<— SIP read from UDP:192.168.3.1:4849 —>
<------------->
<— SIP read from UDP:192.168.3.1:42930 —>
<------------->
<— SIP read from UDP:192.168.3.1:42930 —>
<------------->
<— SIP read from UDP:192.168.3.1:4849 —>
<------------->
<— SIP read from UDP:192.168.3.1:42930 —>
<------------->
*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
Mikrotik
/ip firewall nat
add action=dst-nat chain=dstnat dst-port=10000-20000 log=yes protocol=udp to-addresses=192.168.3.15 to-ports=10000-20000
add action=dst-nat chain=dstnat dst-port=4849 log=yes protocol=udp to-addresses=192.168.3.15 to-ports=4849
add action=dst-nat chain=dstnat dst-port=8288 protocol=tcp to-addresses=192.168.3.15
sip.conf
;
; RTP Configuration
;
[general]
;
; RTP start and RTP end configure start and end addresses
;
; Defaults are rtpstart=5000 and rtpend=31000
;
rtpstart=10000
rtpend=20000
;
; Whether to enable or disable UDP checksums on RTP traffic
;
;rtpchecksums=no
;
; The amount of time a DTMF digit with no ‘end’ marker should be
; allowed to continue (in ‘samples’, 1/8000 of a second)
;
;dtmftimeout=3000
; rtcpinterval = 5000 ; Milliseconds between rtcp reports
;(min 500, max 60000, default 5000)
;
; Enable strict RTP protection. This will drop RTP packets that do not come
; from the recoginized source of the RTP stream. Strict RTP qualifies RTP
; packet stream sources before accepting them upon initial connection and
; when the connection is renegotiated (e.g., transfers and direct media).
; Initial connection and renegotiation starts a learning mode to qualify
; stream source addresses. Once Asterisk has recognized a stream it will
; allow other streams to qualify and replace the current stream for 5
; seconds after starting learning mode. Once learning mode completes the
; current stream is locked in and cannot change until the next
; renegotiation.
; This option is enabled by default.
; strictrtp=yes
;
; Number of packets containing consecutive sequence values needed
; to change the RTP source socket address. This option only comes
; into play while using strictrtp=yes. Consider changing this value
; if rtp packets are dropped from one or both ends after a call is
; connected. This option is set to 4 by default.
; probation=8
;
; Whether to enable or disable ICE support. This option is enabled by default.
; icesupport=false
;
; Hostname or address for the STUN server used when determining the external
; IP address and port an RTP session can be reached at. The port number is
; optional. If omitted the default value of 3478 will be used. This option is
; disabled by default.
;
; e.g. stundaddr:3478=mystun.server.com
;
; stunaddr=
;
; Some multihomed servers have IP interfaces that cannot reach the STUN
; server specified by stunaddr. Blacklist those interface subnets from
; trying to send a STUN packet to find the external IP address.
; Attempting to send the STUN packet needlessly delays processing incoming
; and outgoing SIP INVITEs because we will wait for a response that can
; never come until we give up on the response.
; * Multiple subnets may be listed.
; * Blacklisting applies to IPv4 only. STUN isn’t needed for IPv6.
; * Blacklisting applies when binding RTP to specific IP addresses and not
; the wildcard 0.0.0.0 address. e.g., A PJSIP endpoint binding RTP to a
; specific address using the bind_rtp_to_media_address and media_address
; options. Or the PJSIP endpoint specifies an explicit transport that binds
; to a specific IP address.
;
; e.g. stun_blacklist = 192.168.1.0/255.255.255.0
; stun_blacklist = 10.32.77.0/255.255.255.0
;
; stun_blacklist =
;
; Hostname or address for the TURN server to be used as a relay. The port
; number is optional. If omitted the default value of 3478 will be used.
; This option is disabled by default.
;
; e.g. turnaddr:34780=myturn.server.com
;
; turnaddr=
;
; Username used to authenticate with TURN relay server.
; turnusername=
;
; Password used to authenticate with TURN relay server.
; turnpassword=
;
; Subnets to exclude from ICE host, srflx and relay discovery. This is useful
; to optimize the ICE process where a system has multiple host address ranges
; and/or physical interfaces and certain of them are not expected to be used
; for RTP. For example, VPNs and local interconnections may not be suitable or
; necessary for ICE. Multiple subnets may be listed. If left unconfigured,
; all discovered host addresses are used.
;
; e.g. ice_blacklist = 192.168.1.0/255.255.255.0
; ice_blacklist = 10.32.77.0/255.255.255.0
;
; ice_blacklist =
;
[ice_host_candidates]
;
; When Asterisk is behind a static one-to-one NAT and ICE is in use, ICE will
; expose the server’s internal IP address as one of the host candidates.
; Although using STUN (see the ‘stunaddr’ configuration option) will provide a
; publicly accessible IP, the internal IP will still be sent to the remote
; peer. To help hide the topology of your internal network, you can override
; the host candidates that Asterisk will send to the remote peer.
;
; IMPORTANT: Only use this functionality when your Asterisk server is behind a
; one-to-one NAT and you know what you’re doing. If you do define anything
; here, you almost certainly will NOT want to specify ‘stunaddr’ or ‘turnaddr’
; above.
;
; The format for these overrides is:
;
; =>
;
; The following will replace 192.168.1.10 with 1.2.3.4 during ICE
; negotiation:
;
;192.168.1.10 => 1.2.3.4
;
; You can define an override for more than 1 interface if you have a multihomed
; server. Any local interface that is not matched will be passed through
; unaltered. Both IPv4 and IPv6 addresses are supported.