Hello,
I don’t understand why I have no audio on my endpoints, When I log my softphones from a local address, everything works fine, but if I log from outside my network I have no audio.
I don’t understand how Astertisk can match the local IP address of my client, since In my softphone code I use my public IP address to log to asterisk
I’m using WebRTC for my sofphones
Here is my config
;====================== registration =======
[transport-udp-nat]
type=transport
protocol=udp
bind=0.0.0.0
;local_net=172.16.1.0/32
;local_net=127.0.0.1/32
;external_media_address=41.111.136.198
;external_signaling_address=41.111.136.198
[icosnet]
type=registration
transport=transport-udp-nat
outbound_auth=icosnet
server_uri=sip:196.41.228.32
client_uri=sip:username@196.41.228.32
contact_user=username
retry_interval=60
;from_user=213982420090
;from_domain=196.41.228.32
[icosnet]
type=auth
auth_type=userpass
password=password
username=username
[icosnet]
type=endpoint
context=inbound-calls
disallow=all
allow=ulaw
outbound_auth=icosnet
aors=icosnet
direct_media=no
;======================= Clients =======
[transport-wss]
type=transport
protocol=wss
bind=172.16.1.12
;================= 5099 ===========================================================
[5099]
type=aor
max_contacts=2
remove_existing=yes
[5099]
type=auth
auth_type=userpass
username=5099
password=password
[5099]
type=endpoint
aors=5099
auth=5099
dtls_auto_generate_cert=yes
webrtc=yes
context=default
disallow=all
allow=ulaw
direct_media=no
and this is the Log
<--- Transmitting SIP request (410 bytes) to UDP:196.41.228.32:5060 --->
ACK sip:0667121507@196.41.228.32 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.12:5060;rport;branch=z9hG4bKPjbd88f4fe-256d-4de4-802a-a37563205ca6
From: <sip:5099@172.16.1.12>;tag=248fc000-90b7-4830-83b7-12e2587b525b
To: <sip:0667121507@196.41.228.32>;tag=SIX3Q5Y7TWLSVZDY6EGA____.i
Call-ID: 0f2054e3-6b2c-4999-8d46-1f3aef1bb350
CSeq: 1856 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.4.0
Content-Length: 0
<--- Transmitting SIP request (1107 bytes) to UDP:196.41.228.32:5060 --->
INVITE sip:0667121507@196.41.228.32 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.12:5060;rport;branch=z9hG4bKPjf91e6807-fe96-481b-977c-af42457a0402
From: <sip:5099@172.16.1.12>;tag=248fc000-90b7-4830-83b7-12e2587b525b
To: <sip:0667121507@196.41.228.32>
Contact: <sip:asterisk@172.16.1.12:5060>
Call-ID: 0f2054e3-6b2c-4999-8d46-1f3aef1bb350
CSeq: 1857 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.4.0
Authorization: Digest username="213982420090", realm="sip-1.pb", nonce="1634556467:bbcf374c59d08f2a1c888587e07e44482274ee47", uri="sip:0667121507@196.41.228.32", response="3a5f96f03ae2c4938adf2013915e3105"
Content-Type: application/sdp
Content-Length: 235
v=0
o=- 1707974439 1707974439 IN IP4 172.16.1.12
s=Asterisk
c=IN IP4 172.16.1.12
t=0 0
m=audio 11828 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<--- Received SIP response (521 bytes) from UDP:196.41.228.32:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.1.12:5060;rport=65476;branch=z9hG4bKPjf91e6807-fe96-481b-977c-af42457a0402;received=41.111.136.198
Record-Route: <sip:196.41.228.32;lr;ep;pinhole=UDP:41.111.136.198:65476>
To: <sip:0667121507@196.41.228.32>
From: <sip:5099@172.16.1.12>;tag=248fc000-90b7-4830-83b7-12e2587b525b
Call-ID: 0f2054e3-6b2c-4999-8d46-1f3aef1bb350
CSeq: 1857 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, UPDATE
Server: PortaSIP
Content-Length: 0
<--- Received SIP response (714 bytes) from UDP:196.41.228.32:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.1.12:5060;rport=65476;branch=z9hG4bKPjf91e6807-fe96-481b-977c-af42457a0402;received=41.111.136.198
Record-Route: <sip:196.41.228.32;lr;ep;pinhole=UDP:41.111.136.198:65476>
Contact: sip:196.41.228.115:5070
To: <sip:0667121507@196.41.228.32>;tag=SIX3Q5Y7TWLSVZDY6PFQ____.i
From: <sip:5099@172.16.1.12>;tag=248fc000-90b7-4830-83b7-12e2587b525b
Call-ID: 0f2054e3-6b2c-4999-8d46-1f3aef1bb350
CSeq: 1857 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, UPDATE
Server: PortaSIP
Portasip-cps-limit: max=1;method=normal;aggregation-id=i_account:1431877;matched-by=digest_username;matched-value=213982420090
Content-Length: 0
<--- Received SIP response (993 bytes) from UDP:196.41.228.32:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.1.12:5060;rport=65476;branch=z9hG4bKPjf91e6807-fe96-481b-977c-af42457a0402;received=41.111.136.198
Record-Route: <sip:196.41.228.32;lr;ep;pinhole=UDP:41.111.136.198:65476>
Contact: sip:196.41.228.115:5070
To: <sip:0667121507@196.41.228.32>;tag=SIX3Q5Y7TWLSVZDY6PFQ____.i
From: <sip:5099@172.16.1.12>;tag=248fc000-90b7-4830-83b7-12e2587b525b
Call-ID: 0f2054e3-6b2c-4999-8d46-1f3aef1bb350
CSeq: 1857 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, UPDATE
Content-Type: application/sdp
Server: PortaSIP
Portasip-cps-limit: max=1;method=normal;aggregation-id=i_account:1431877;matched-by=digest_username;matched-value=213982420090
Content-Length: 250
v=0
o=PortaSIP 3075015875196607957 1 IN IP4 196.41.228.115
s=-
t=0 0
m=audio 56002 RTP/AVP 0 101
c=IN IP4 196.41.228.115
b=AS:128
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:40
a=sendrecv
> 0x7fb6d8031960 -- Strict RTP learning after remote address set to: 196.41.228.115:56002
<--- Transmitting SIP request (472 bytes) to UDP:196.41.228.32:5060 --->
ACK sip:196.41.228.115:5070 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.12:5060;rport;branch=z9hG4bKPj9b6abdd4-7be6-43f7-aaee-57fa0603736d
From: <sip:5099@172.16.1.12>;tag=248fc000-90b7-4830-83b7-12e2587b525b
To: <sip:0667121507@196.41.228.32>;tag=SIX3Q5Y7TWLSVZDY6PFQ____.i
Call-ID: 0f2054e3-6b2c-4999-8d46-1f3aef1bb350
CSeq: 1857 ACK
Route: <sip:196.41.228.32;lr;ep;pinhole=UDP:41.111.136.198:65476>
Max-Forwards: 70
User-Agent: Asterisk PBX 18.4.0
Content-Length: 0
-- PJSIP/icosnet-00000001 answered PJSIP/5099-00000000
> 0x7fb6d8052150 -- Strict RTP learning after remote address set to: 192.168.163.245:62400
<--- Transmitting SIP response (1553 bytes) to WSS:105.235.129.145:25030 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS l9oh3rb05iga.invalid;rport=25030;received=105.235.129.145;branch=z9hG4bK7137964
Call-ID: drd9g7t6hd1hdtl8gs9h
From: <sip:5099@41.111.136.198>;tag=q2nvc83tg0
To: <sip:0667121507@41.111.136.198>;tag=5bb30772-4faa-4b86-be01-81c8e063d5eb
CSeq: 9480 INVITE
Server: Asterisk PBX 18.4.0
Contact: <sip:172.16.1.12:1983;transport=ws>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 90;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length: 922
v=0
o=- 897551239 2 IN IP4 172.16.1.12
s=Asterisk
c=IN IP4 172.16.1.12
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0
m=audio 16760 UDP/TLS/RTP/SAVPF 0 101
a=connection:new
a=setup:active
a=fingerprint:SHA-256 1F:3D:07:84:51:7B:93:80:9B:C8:3B:95:8C:09:51:98:43:22:EA:47:52:0C:65:32:71:95:D7:26:A5:21:31:C3
a=ice-ufrag:5e4fbc172d1a49524aa0b5192ad4827b
a=ice-pwd:019e8ed864764ea37134d924337b4b7b
a=candidate:Hac10010c 1 UDP 2130706431 172.16.1.12 16760 typ host
a=candidate:Hac110001 1 UDP 2130706431 172.17.0.1 16760 typ host
a=candidate:Hc824934 1 UDP 2130706431 fe80::e12e:53dc:1cf7:e6ff 16760 typ host
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp-mux
a=ssrc:271470930 cname:15993d2a-f0be-486f-a2f5-42ac64b552d0
a=msid:54008c26-3c41-4efa-84da-d12d9d449229 c677551e-024c-4adb-897b-412e220acf6a
a=rtcp-fb:* transport-cc
a=mid:0
-- Channel PJSIP/icosnet-00000001 joined 'simple_bridge' basic-bridge <a9cb77b7-301a-4348-bf5a-15ed933e4318>
-- Channel PJSIP/5099-00000000 joined 'simple_bridge' basic-bridge <a9cb77b7-301a-4348-bf5a-15ed933e4318>
<--- Received SIP request (437 bytes) from WSS:105.235.129.145:25030 --->
ACK sip:172.16.1.12:1983;transport=ws SIP/2.0
Via: SIP/2.0/WSS l9oh3rb05iga.invalid;branch=z9hG4bK7384895
Max-Forwards: 69
To: <sip:0667121507@41.111.136.198>;tag=5bb30772-4faa-4b86-be01-81c8e063d5eb
From: <sip:5099@41.111.136.198>;tag=q2nvc83tg0
Call-ID: drd9g7t6hd1hdtl8gs9h
CSeq: 9480 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: outbound
User-Agent: JsSIP 3.8.0
Content-Length: 0
> 0x7fb6d8052150 -- Strict RTP learning after ICE completion
<--- Received SIP request (481 bytes) from WSS:105.235.129.145:25030 --->
BYE sip:172.16.1.12:1983;transport=ws SIP/2.0
Via: SIP/2.0/WSS l9oh3rb05iga.invalid;branch=z9hG4bK5415583
Max-Forwards: 69
To: <sip:0667121507@41.111.136.198>;tag=5bb30772-4faa-4b86-be01-81c8e063d5eb
From: <sip:5099@41.111.136.198>;tag=q2nvc83tg0
Call-ID: drd9g7t6hd1hdtl8gs9h
CSeq: 9481 BYE
Reason: SIP ;cause=408; text="RTP Timeout"
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: outbound
User-Agent: JsSIP 3.8.0
Content-Length: 0
<--- Transmitting SIP response (338 bytes) to WSS:105.235.129.145:25030 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS l9oh3rb05iga.invalid;rport=25030;received=105.235.129.145;branch=z9hG4bK5415583
Call-ID: drd9g7t6hd1hdtl8gs9h
From: <sip:5099@41.111.136.198>;tag=q2nvc83tg0
To: <sip:0667121507@41.111.136.198>;tag=5bb30772-4faa-4b86-be01-81c8e063d5eb
CSeq: 9481 BYE
Server: Asterisk PBX 18.4.0
Content-Length: 0