I have two ways audio when calling from a WebRTC client connected to Asterisk to my mobile phone, but when I call from the mobile phone to the WebRTC client the call is established and there is only one way audio: from the WebRTC client to the mobile phone.
I'm using Asterisk 15.6.1 installed on a VPS with static IP, the WebRTC client is a browser softphone using the SIP.js library, and I have a local phone number from Localphone.
My "pjsip.conf" relevant settings are:
[localphone]
type=registration
transport=transport-udp
outbound_auth=localphone
client_uri = sip:[email protected]:5060
server_uri = sip:localphone.com:5060
auth_rejection_permanent=no
contact_user=12345678
[localphone]
type=auth
auth_type=userpass
username=12345678
password=mypassword
[localphone]
type=aor
max_contacts=100
contact=sip:[email protected]
[localphone]
type=endpoint
transport=transport-udp
context=localphone-inc
disallow = all
allow = ulaw
allow = alaw
rtcp_mux=yes
ice_support=yes
direct_media=no
from_user=12345678
from_domain=localphone.com
outbound_auth=localphone
aors=localphone
[localphone]
type = identify
endpoint = localphone
match = 140.153.72.56
; This is the WebRTC client
[1652]
type=aor
max_contacts=100
[auth1652]
type=auth
auth_type=userpass
username=1652
password=complicatedpassword
[1652]
type=endpoint
context=localphone-pjsip
aors=1652
auth=auth1652
transport=transport-wss
webrtc=yes
disallow=all
allow=ulaw
allow=alaw
dtls_cert_file=/etc/asterisk/key/asterisk.pem
dtls_private_key=/etc/asterisk/key/asterisk.key
[1652]
type = identify
endpoint = 1652
match = 123.123.123.123 ; the public IP of the VPS
The Asterisk debug log doesn’t show errors. The SIP session looks like this:
<--- Received SIP request (1175 bytes) from UDP:140.153.72.56:5060 --->
INVITE sip:136.46.324.75:5060 SIP/2.0
Record-Route: <sip:140.153.72.56;lr=on;ftag=gK086bb5a7>
Record-Route: <sip:126.219.52.41;lr;ftag=gK086bb5a7>
Via: SIP/2.0/UDP 140.153.72.56;branch=z9hG4bK356c.5360dbb7.0
Via: SIP/2.0/UDP 126.219.52.41:5060;rport=5060;branch=z9hG4bK356c.10463da.0
From: <sip:[email protected]>;tag=gK086bb5a7
To: <sip:[email protected]>;tag=514dc3ba-68ed-42ff-9733-bd65b8ebb7c7
Call-ID: [email protected]
CSeq: 63984 INVITE
Max-Forwards: 12
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:126.219.52.41;vbdid=941.29056c13>
Supported: timer
Session-Expires: 1800;refresher=uac
Min-SE: 90
Content-Length: 239
Content-Disposition: session; handling=required
Content-Type: application/sdp
v=0
o=Sonus_UAC 13229 650067 IN IP4 199.199.12.56
s=SIP Media Capabilities
c=IN IP4 199.199.12.54
t=0 0
m=audio 40808 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=sendrecv
a=maxptime:20
> 0x7fddf001b170 -- Strict RTP learning after remote address set to: 199.199.12.54:40808
<--- Transmitting SIP response (1062 bytes) to UDP:140.153.72.56:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 140.153.72.56;rport=5060;received=140.153.72.56;branch=z9hG4bK356c.5360dbb7.0
Via: SIP/2.0/UDP 126.219.52.41:5060;rport=5060;branch=z9hG4bK356c.10463da.0
Record-Route: <sip:140.153.72.56;lr;ftag=gK086bb5a7>
Record-Route: <sip:126.219.52.41;lr;ftag=gK086bb5a7>
Call-ID: [email protected]
From: <sip:[email protected]>;tag=gK086bb5a7
To: <sip:[email protected]>;tag=514dc3ba-68ed-42ff-9733-bd65b8ebb7c7
CSeq: 63984 INVITE
Session-Expires: 1800;refresher=uac
Require: timer
Contact: <sip:136.46.324.75:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Server: Asterisk PBX 15.6.1
Content-Type: application/sdp
Content-Length: 234
v=0
o=- 13229 650069 IN IP4 136.46.324.75
s=Asterisk
c=IN IP4 136.46.324.75
t=0 0
m=audio 10010 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
> 0x7fddf001b170 -- Strict RTP switching to RTP target address 199.199.12.54:40808 as source
<--- Received SIP request (411 bytes) from UDP:140.153.72.56:5060 --->
ACK sip:136.46.324.75:5060 SIP/2.0
Via: SIP/2.0/UDP 140.153.72.56;branch=z9hG4bK356c.5360dbb7.2
Via: SIP/2.0/UDP 126.219.52.41:5060;rport=5060;branch=z9hG4bK356c.10463da.2
From: <sip:[email protected]>;tag=gK086bb5a7
To: <sip:[email protected]>;tag=514dc3ba-68ed-42ff-9733-bd65b8ebb7c7
Call-ID: [email protected]
CSeq: 63984 ACK
Max-Forwards: 12
Content-Length: 0
<--- Received SIP request (428 bytes) from WSS:152.231.162.25:53283 --->
BYE sip:[email protected]:5060;transport=ws SIP/2.0
Via: SIP/2.0/WSS qae9g3ug98cm.invalid;branch=z9hG4bK8418203
Max-Forwards: 70
To: <sip:[email protected]>;tag=46b408fd-815a-49c1-b337-74ee894d6e44
From: "Grant Brandon" <sip:[email protected]>;tag=1l7sldm3o2
Call-ID: 4b8d773b-243b-4e97-9962-efc4d55eb0de
CSeq: 27946 BYE
Supported: outbound
User-Agent: SIP.js/0.7.8
Content-Length: 0
<--- Transmitting SIP response (376 bytes) to WSS:152.231.162.25:53283 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS qae9g3ug98cm.invalid;rport=53283;received=152.231.162.25;branch=z9hG4bK8418203
Call-ID: 4b8d773b-243b-4e97-9962-efc4d55eb0de
From: "Grant Brandon" <sip:[email protected]>;tag=1l7sldm3o2
To: <sip:[email protected]>;tag=46b408fd-815a-49c1-b337-74ee894d6e44
CSeq: 27946 BYE
Server: Asterisk PBX 15.6.1
Content-Length: 0
-- Channel PJSIP/601-00000003 left 'simple_bridge' basic-bridge <f59afd46-1e16-4f47-9a9d-59c11987cc77>
-- Channel PJSIP/localphone-00000002 left 'simple_bridge' basic-bridge <f59afd46-1e16-4f47-9a9d-59c11987cc77>
== Spawn extension (localphone-pjsip, 601, 1) exited non-zero on 'PJSIP/localphone-00000002'
<--- Transmitting SIP request (487 bytes) to UDP:140.153.72.56:5060 --->
BYE sip:126.219.52.41;vbdid=941.29056c13 SIP/2.0
Via: SIP/2.0/UDP 136.46.324.75:5060;rport;branch=z9hG4bKPjf05b6f33-c9d2-4606-80d5-1e54fc110d40
From: <sip:[email protected]>;tag=514dc3ba-68ed-42ff-9733-bd65b8ebb7c7
To: <sip:[email protected]>;tag=gK086bb5a7
Call-ID: [email protected]
CSeq: 13266 BYE
Route: <sip:140.153.72.56;lr;ftag=gK086bb5a7>
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 15.6.1
Content-Length: 0
<--- Received SIP response (335 bytes) from UDP:140.153.72.56:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 136.46.324.75:5060;rport=5060;branch=z9hG4bKPjf05b6f33-c9d2-4606-80d5-1e54fc110d40
From: <sip:[email protected]>;tag=514dc3ba-68ed-42ff-9733-bd65b8ebb7c7
To: <sip:[email protected]>;tag=gK086bb5a7
Call-ID: [email protected]
CSeq: 13266 BYE
Content-Length: 0
Your help will be appreciated !
