Asterisk doesn't work completely with WEBRTC

Asterisk Asterisk WebRTC
1

Hello,

I have been trying for several weeks to make calls from a SIP client (Linphone) to a WEB client which is SIPML5 but without success. However, calling the other way works perfectly.
To be more precise when a call is launched from Linphone to SIPML5 we can see that it hangs up immediately. On the side of the browser where SIPML5 is opened (Firefox or Chrome) we can see that no request arrives.

Here is the SIPML5 configuration:
image

sipml5_expert-mode
sipml5_expert-mode752×541 48.4 KB

Here is the Asterisk 18.2.2 (In a Docker under Linux) configuration:

             - extensions.conf:

[parking]
exten => 216,1,Wait(1)
exten => 216,2,Answer
exten => 216,3,Dial(PJSIP/216)
exten => 216,4,Hangup

exten => webrtc_client,1,Wait(1)
exten => webrtc_client,2,Dial(PJSIP/webrtc_client)
exten => webrtc_client,3,Hangup


          - pjsip.conf:

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0

[216]
type=endpoint
transport=transport-udp
context=parking
use_avpf=yes
media_encryption=sdes
media_use_received_transport=yes
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=g722
auth=216
aors=216
callerid=216<parking>

[216]
type=auth
auth_type=userpass
password=***********
username=216

[216]
type=aor
max_contacts=1

[transport-wss]            
type=transport              
protocol=wss               
bind=0.0.0.0                
                         
[webrtc_client]          
type=aor                 
max_contacts=5                  
remove_existing=yes                            
                                       
[webrtc_client]          
type=auth                
auth_type=userpass 
username=webrtc_client
password=******************
                                                                                    
[webrtc_client]                                                                     
type=endpoint                                                                       
transport=transport-wss                                                                      
webrtc=yes                                                                                                                                              
context=parking                                                                     
disallow=all                                                                                                                                                 
allow=ulaw                                                           
allow=alaw                                                           
allow=g729                                                           
allow=g722                                                           
auth=webrtc_client                                                   
aors=webrtc_client                                                   
callerid=webrtc<parking>          

        -http.conf: 

[general]                                                                                                                                
servername=Asterisk                                                                                                                                                         
enabled=yes                                                                                                                                                                               
bindaddr=0.0.0.0                                                                                                                                                                            
bindport=8088                                                                                                                                                 
prefix=asterisk     
sessionlimit=100                                                                                                                                          
enablestatic=yes                                                                                                                                      
redirect = / /static/dev_interface_Asterisk/edition.html                                                                            
tlsenable=yes                         
tlsbindaddr=0.0.0.0:8089                                                                    
tlscertfile=/etc/asterisk/keys/asterisk.pem 
tlsprivatekey=/etc/asterisk/keys/asterisk.key

And finally here is the trace with the part containing the error during a call test:


[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004: Stream 0:audio-0:audio:sendrecv (ulaw|alaw|g729|g722) added with mid audio-0
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004: Done with 0:audio-0:audio:sendrecv (ulaw|alaw|g729|g722)
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004: Adding bundle groups (if available)
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004: Copying connection details
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004: Processing media 0
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004: Media 0 reset
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004: Method is INVITE
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004
[Sep 27 11:18:50] DEBUG[21655] res_pjsip/pjsip_resolver.c: Performing SIP DNS resolution of target '172.21.38.236'
[Sep 27 11:18:50] DEBUG[21655] res_pjsip/pjsip_resolver.c: Transport type for target '172.21.38.236' is '(null)'
[Sep 27 11:18:50] DEBUG[21655] res_pjsip/pjsip_resolver.c: Target '172.21.38.236' is an IP address, skipping resolution
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004 Event: TSX_STATE  Inv State: DISCONNCTD
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: Function session_inv_on_state_changed called on event TSX_STATE
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The state change pertains to the endpoint 'webrtc_client(PJSIP/webrtc_client-00000004)'
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The inv session still has an invite_tsx (0x7f02db5db0c8)
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: There is no transaction involved in this state change
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The current inv state is DISCONNCTD
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: PJSIP/webrtc_client-00000004: Source of transaction state change is TRANSPORT_ERROR
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004 TSX State: Terminated  Inv State: DISCONNCTD
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: Function session_inv_on_tsx_state_changed called on event TSX_STATE
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The state change pertains to the endpoint 'webrtc_client(PJSIP/webrtc_client-00000004)'
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The inv session does NOT have an invite_tsx
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The UAC INVITE transaction involved in this state change is 0x7f02db5db0c8
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The current transaction state is Terminated
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The transaction state change event is TRANSPORT_ERROR
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The current inv state is DISCONNCTD
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  Disconnected
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  (null session) TSX State: Terminated  Inv State: DISCONNCTD
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  
[Sep 27 11:18:50] DEBUG[21655] chan_pjsip.c:  RC: 0
[Sep 27 11:18:50] DEBUG[21655] chan_pjsip.c:  PJSIP/webrtc_client-00000004
[Sep 27 11:18:50] DEBUG[21655] chan_pjsip.c:  
[Sep 27 11:18:50] DEBUG[21681][C-00000003] channel.c: PJSIP/216-00000003: Dropping redundant connected line update "webrtc" <parking>.
[Sep 27 11:18:50] DEBUG[21681][C-00000003] channel.c: Channel 0x7f02d8cfa080 'PJSIP/webrtc_client-00000004' hanging up.  Refs: 2
[Sep 27 11:18:50] DEBUG[21681][C-00000003] chan_pjsip.c:  PJSIP/webrtc_client-00000004
[Sep 27 11:18:50] DEBUG[21681][C-00000003] chan_pjsip.c:  Cause: 503
[Sep 27 11:18:50] DEBUG[21525] cdr.c: Finalized CDR for PJSIP/216-00000003 - start 1664270329.398984 answer 0.000000 end 1664270330.414431 dur 1.015 bill 1664270330.414 dispo FAILED
[Sep 27 11:18:50] VERBOSE[21681][C-00000003] app_dial.c: Everyone is busy/congested at this time (1:0/1/0)
[Sep 27 11:18:50] DEBUG[21655] chan_pjsip.c:  PJSIP/webrtc_client-00000004
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004 Response 503
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: webrtc_client: Destroying SIP session
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d885b690) DTLS srtp - stopped timeout timer'
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d885b690) DTLS srtp - stopped timeout timer'
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d885b690) DTLS stop
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d885b690) DTLS srtp - stopped timeout timer'
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d885b690) DTLS srtp - stopped timeout timer'
[Sep 27 11:18:50] DEBUG[21681][C-00000003] app_dial.c:  PJSIP/216-00000003: No outging channels available
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d885b690) ICE RTP transport deallocating
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d885b690) ICE stopped
[Sep 27 11:18:50] DEBUG[21681][C-00000003] app_dial.c: Exiting with DIALSTATUS=CONGESTION.
[Sep 27 11:18:50] DEBUG[21655] rtp_engine.c: Destroyed RTP instance '0x7f02d885b690'
[Sep 27 11:18:50] DEBUG[21681][C-00000003] app_dial.c:  PJSIP/216-00000003: Done
[Sep 27 11:18:50] DEBUG[21655] channel.c: Channel 0x7f02d8cfa080 'PJSIP/webrtc_client-00000004' destroying
[Sep 27 11:18:50] DEBUG[21681][C-00000003] pbx.c: Launching 'Hangup'
[Sep 27 11:18:50] DEBUG[21525] cdr.c: Finalized CDR for PJSIP/webrtc_client-00000004 - start 1664270330.405554 answer 0.000000 end 1664270330.416349 dur 0.010 bill 1664270330.416 dispo FAILED
[Sep 27 11:18:50] DEBUG[21525] cdr.c: CDR for PJSIP/webrtc_client-00000004 is dialed and has no Party B; discarding
[Sep 27 11:18:50] DEBUG[21655] stasis.c: Destroying topic. name: channel:1664270330.4, detail: 
[Sep 27 11:18:50] DEBUG[21655] stasis.c: Topic 'channel:1664270330.4': 0x7f02d84a5370 destroyed
[Sep 27 11:18:50] DEBUG[21655] channel_internal_api.c:  <initializing>: MultistreamFormats: (nothing)
[Sep 27 11:18:50] DEBUG[21655] channel_internal_api.c:  Channel is being initialized or destroyed
[Sep 27 11:18:50] DEBUG[21655] chan_pjsip.c:  
[Sep 27 11:18:50] DEBUG[21517] devicestate.c: No provider found, checking channel drivers for PJSIP - webrtc_client
[Sep 27 11:18:50] DEBUG[21517] devicestate.c: Changing state for PJSIP/webrtc_client - state 1 (Not in use)
[Sep 27 11:18:50] VERBOSE[21681][C-00000003] pbx.c: Executing [webrtc_client@parking:3] Hangup("PJSIP/216-00000003", "") in new stack
[Sep 27 11:18:50] DEBUG[21681][C-00000003] channel.c: Soft-Hanging (0x20) up channel 'PJSIP/216-00000003'
[Sep 27 11:18:50] DEBUG[21681][C-00000003] pbx.c: Spawn extension (parking,webrtc_client,3) exited non-zero on 'PJSIP/216-00000003'
[Sep 27 11:18:50] VERBOSE[21681][C-00000003] pbx.c: Spawn extension (parking, webrtc_client, 3) exited non-zero on 'PJSIP/216-00000003'
[Sep 27 11:18:50] DEBUG[21681][C-00000003] channel.c: Soft-Hanging (0x10) up channel 'PJSIP/216-00000003'
[Sep 27 11:18:50] DEBUG[21681][C-00000003] channel.c: Channel 0x7f02db5ab190 'PJSIP/216-00000003' hanging up.  Refs: 2
[Sep 27 11:18:50] DEBUG[21681][C-00000003] chan_pjsip.c:  PJSIP/216-00000003
[Sep 27 11:18:50] DEBUG[21681][C-00000003] chan_pjsip.c:  Cause: 503
[Sep 27 11:18:50] DEBUG[21655] chan_pjsip.c:  PJSIP/216-00000003
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/216-00000003 Response 503
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/216-00000003: Method is INVITE, Response is 503 Service Unavailable
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/216-00000003
[Sep 27 11:18:50] VERBOSE[21655] res_pjsip_logger.c: <--- Transmitting SIP response (369 bytes) to UDP:172.21.38.236:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 172.21.38.236:5060;rport=5060;received=172.21.38.236;branch=z9hG4bK.1jH3Y1hfB
Call-ID: ~-NoF7d-dr
From: "216" <sip:216@172.21.38.222>;tag=yO1RnUwCh
To: <sip:webrtc_client@172.21.38.222>;tag=aFKykoc3031flZQBDIhefL5wEf59WXT5
CSeq: 21 INVITE
Server: Asterisk PBX 18.2.2
Reason: Q.850;cause=34
Content-Length:  0

After long days of research on the internet (forum, website …) I do not know what to look for.
If you need more precision do not hesitate.
Thank you for your interest.

2

You can find the complete trace here:

[Sep 27 11:18:49] VERBOSE[21534] res_pjsip_logger.c: <--- Received SIP request (1753 bytes) from UDP:172.21.38.236:5060 --->
INVITE sip:webrtc_client@172.21.38.222 SIP/2.0
Via: SIP/2.0/UDP 172.21.38.236:5060;branch=z9hG4bK.zfCyj54m1;rport
From: "216" <sip:216@172.21.38.222>;tag=yO1RnUwCh
To: sip:webrtc_client@172.21.38.222
CSeq: 20 INVITE
Call-ID: ~-NoF7d-dr
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 1085
Contact: <sip:216@172.21.38.236;transport=udp>;expires=3599;+sip.instance="<urn:uuid:e2c6096f-11a0-0061-a97d-e2c741d445dd>"
User-Agent: Linphone Desktop/4.2.5 (Windows 10 Version 1607, Qt 5.14.2) LinphoneCore/4.4.19

v=0
o=216 1167 3598 IN IP4 172.21.38.236
s=Talk
c=IN IP4 172.21.38.236
t=0 0
a=ice-pwd:2409854524fd4d0482bbfabc
a=ice-ufrag:a73358af
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP[Sep 27 11:18:49] DEBUG[21534] res_pjsip/pjsip_distributor.c: Could not find matching transaction for Request msg INVITE/cseq=20 (rdata0x7f02db5bb0c8)
[Sep 27 11:18:49] DEBUG[21534] res_pjsip/pjsip_distributor.c: Calculated serializer pjsip/distributor-00000039 to use for Request msg INVITE/cseq=20 (rdata0x7f02db5bb0c8)
[Sep 27 11:18:49] DEBUG[21655] netsock2.c: Splitting '172.21.38.236' into...
[Sep 27 11:18:49] DEBUG[21655] netsock2.c: ...host '172.21.38.236' and port ''.
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_endpoint_identifier_ip.c: No identify sections to match against
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_endpoint_identifier_user.c: Attempting identify by From username '216' domain '172.21.38.222'
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_endpoint_identifier_user.c: Identified by From username '216' domain '172.21.38.222'
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_authenticator_digest.c: Using default realm 'asterisk' on incoming auth '216'.
[Sep 27 11:18:49] DEBUG[21655] netsock2.c: Splitting '172.21.38.222' into...
[Sep 27 11:18:49] DEBUG[21655] netsock2.c: ...host '172.21.38.222' and port ''.
[Sep 27 11:18:49] DEBUG[21655] netsock2.c: Splitting '172.21.38.236' into...
[Sep 27 11:18:49] DEBUG[21655] netsock2.c: ...host '172.21.38.236' and port ''.
[Sep 27 11:18:49] VERBOSE[21655] res_pjsip_logger.c: <--- Transmitting SIP response (469 bytes) to UDP:172.21.38.236:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.21.38.236:5060;rport=5060;received=172.21.38.236;branch=z9hG4bK.zfCyj54m1
Call-ID: ~-NoF7d-dr
From: "216" <sip:216@172.21.38.222>;tag=yO1RnUwCh
To: <sip:webrtc_client@172.21.38.222>;tag=z9hG4bK.zfCyj54m1
CSeq: 20 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1664270329/a235b7d893668e09e52250c80a11a4a1",opaque="38f18ea31440d460",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.2.2
Content-Length:  0


[Sep 27 11:18:49] VERBOSE[21534] res_pjsip_logger.c: <--- Received SIP request (405 bytes) from UDP:172.21.38.236:5060 --->
ACK sip:webrtc_client@172.21.38.222 SIP/2.0
Via: SIP/2.0/UDP 172.21.38.236:5060;branch=z9hG4bK.zfCyj54m1;rport
Call-ID: ~-NoF7d-dr
From: "216" <sip:216@172.21.38.222>;tag=yO1RnUwCh
To: <sip:webrtc_client@172.21.38.222>;tag=z9hG4bK.zfCyj54m1
Contact: <sip:216@172.21.38.236;transport=udp>;expires=3599;+sip.instance="<urn:uuid:e2c6096f-11a0-0061-a97d-e2c741d445dd>"
Max-Forwards: 70
CSeq: 20 ACK


[Sep 27 11:18:49] DEBUG[21534] res_pjsip/pjsip_distributor.c: Could not find matching transaction for Request msg ACK/cseq=20 (rdata0x7f02db5ec078)
[Sep 27 11:18:49] DEBUG[21534] res_pjsip/pjsip_distributor.c: Calculated serializer pjsip/distributor-00000039 to use for Request msg ACK/cseq=20 (rdata0x7f02db5ec078)
[Sep 27 11:18:49] DEBUG[21655] netsock2.c: Splitting '172.21.38.236' into...
[Sep 27 11:18:49] DEBUG[21655] netsock2.c: ...host '172.21.38.236' and port ''.
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_endpoint_identifier_ip.c: No identify sections to match against
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_endpoint_identifier_user.c: Attempting identify by From username '216' domain '172.21.38.222'
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_endpoint_identifier_user.c: Identified by From username '216' domain '172.21.38.222'
[Sep 27 11:18:49] VERBOSE[21534] res_pjsip_logger.c: <--- Received SIP request (2040 bytes) from UDP:172.21.38.236:5060 --->
INVITE sip:webrtc_client@172.21.38.222 SIP/2.0
Via: SIP/2.0/UDP 172.21.38.236:5060;branch=z9hG4bK.1jH3Y1hfB;rport
From: "216" <sip:216@172.21.38.222>;tag=yO1RnUwCh
To: sip:webrtc_client@172.21.38.222
CSeq: 21 INVITE
Call-ID: ~-NoF7d-dr
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 1085
Contact: <sip:216@172.21.38.236;transport=udp>;expires=3599;+sip.instance="<urn:uuid:e2c6096f-11a0-0061-a97d-e2c741d445dd>"
User-Agent: Linphone Desktop/4.2.5 (Windows 10 Version 1607, Qt 5.14.2) LinphoneCore/4.4.19
Authorization:  Digest realm="asterisk", nonce="1664270329/a235b7d893668e09e52250c80a11a4a1", algorithm=md5, opaque="38f18ea31440d460", username="216",  uri="sip:webrtc_client@172.21.38.222", response="0f422e2749eb52950f75e33e9f80b1[Sep 27 11:18:49] DEBUG[21534] res_pjsip/pjsip_distributor.c: Could not find matching transaction for Request msg INVITE/cseq=21 (rdata0x7f02db5fd098)
[Sep 27 11:18:49] DEBUG[21534] res_pjsip/pjsip_distributor.c: Calculated serializer pjsip/distributor-00000039 to use for Request msg INVITE/cseq=21 (rdata0x7f02db5fd098)
[Sep 27 11:18:49] DEBUG[21655] netsock2.c: Splitting '172.21.38.236' into...
[Sep 27 11:18:49] DEBUG[21655] netsock2.c: ...host '172.21.38.236' and port ''.
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_endpoint_identifier_ip.c: No identify sections to match against
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_endpoint_identifier_user.c: Attempting identify by From username '216' domain '172.21.38.222'
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_endpoint_identifier_user.c: Identified by From username '216' domain '172.21.38.222'
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_authenticator_digest.c: Using default realm 'asterisk' on incoming auth '216'.
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_authenticator_digest.c: Calculated nonce 1664270329/a235b7d893668e09e52250c80a11a4a1. Actual nonce is 1664270329/a235b7d893668e09e52250c80a11a4a1
[Sep 27 11:18:49] DEBUG[21655] netsock2.c: Splitting '172.21.38.222' into...
[Sep 27 11:18:49] DEBUG[21655] netsock2.c: ...host '172.21.38.222' and port ''.
[Sep 27 11:18:49] DEBUG[21655] netsock2.c: Splitting '172.21.38.236' into...
[Sep 27 11:18:49] DEBUG[21655] netsock2.c: ...host '172.21.38.236' and port ''.
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  (null session) Request: INVITE sip:webrtc_client@172.21.38.222
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  Request: sip:webrtc_client@172.21.38.222
[Sep 27 11:18:49] DEBUG[21655] res_pjsip/pjsip_distributor.c: Calculated serializer pjsip/distributor-00000039 to use for Request msg INVITE/cseq=21 (rdata0x7f02db5ec058)
[Sep 27 11:18:49] DEBUG[21655] chan_pjsip.c:  216
[Sep 27 11:18:49] DEBUG[21655] chan_pjsip.c:  Direct media no glare mitigation
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216: Call (UDP:172.21.38.236:5060) to extension 'webrtc_client' sending 100 Trying
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216: Method is INVITE, Response is 100 Trying
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216
[Sep 27 11:18:49] VERBOSE[21655] res_pjsip_logger.c: <--- Transmitting SIP response (295 bytes) to UDP:172.21.38.236:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.21.38.236:5060;rport=5060;received=172.21.38.236;branch=z9hG4bK.1jH3Y1hfB
Call-ID: ~-NoF7d-dr
From: "216" <sip:216@172.21.38.222>;tag=yO1RnUwCh
To: <sip:webrtc_client@172.21.38.222>
CSeq: 21 INVITE
Server: Asterisk PBX 18.2.2
Content-Length:  0


[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216 Event: TSX_STATE  Inv State: INCOMING
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c: Function session_inv_on_state_changed called on event TSX_STATE
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c: The state change pertains to the endpoint '216()'
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c: The inv session still has an invite_tsx (0x7f02db5dccc8)
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c: There is no transaction involved in this state change
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c: The current inv state is INCOMING
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c: 216: Source of transaction state change is TX_MSG
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216 TSX State: Proceeding  Inv State: INCOMING
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c: Function session_inv_on_tsx_state_changed called on event TSX_STATE
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c: The state change pertains to the endpoint '216()'
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c: The inv session still has an invite_tsx (0x7f02db5dccc8)
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c: The UAS INVITE transaction involved in this state change is 0x7f02db5dccc8
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c: The current transaction state is Proceeding
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c: The transaction state change event is TX_MSG
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c: The current inv state is INCOMING
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  Nothing delayed
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216 TSX State: Proceeding  Inv State: INCOMING
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  Topology: Pending: (null topology)  Active: (null topology)
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216: Media count: 1
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216: Processing stream 0
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216: Using audio-0 for new stream name
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216: Using new stream 0:audio-0:audio:sendrecv (nothing)
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216 Adding position 0
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  Creating new media session
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  Setting media session as default for audio
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  Done
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216: Negotiating incoming SDP media stream 0:audio-0:audio:sendrecv (nothing) using audio SDP handler
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_sdp_rtp.c:  216
[Sep 27 11:18:49] DEBUG[21655] netsock2.c: Splitting '172.21.38.236' into...
[Sep 27 11:18:49] DEBUG[21655] netsock2.c: ...host '172.21.38.236' and port ''.
[Sep 27 11:18:49] DEBUG[21655] netsock2.c: Splitting '0.0.0.0' into...
[Sep 27 11:18:49] DEBUG[21655] netsock2.c: ...host '0.0.0.0' and port ''.
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_sdp_rtp.c: Transport transport-udp bound to 0.0.0.0: Using it for RTP media.
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f02d8858f20'
[Sep 27 11:18:49] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d8858f20) RTP allocated port 14378
[Sep 27 11:18:49] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d8858f20) ICE creating session 0.0.0.0:14378 (14378)
[Sep 27 11:18:49] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d8858f20) ICE create
[Sep 27 11:18:49] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d8858f20) ICE add system candidates
[Sep 27 11:18:49] DEBUG[21655] netsock2.c: Splitting '172.21.38.222' into...
[Sep 27 11:18:49] DEBUG[21655] netsock2.c: ...host '172.21.38.222' and port ''.
[Sep 27 11:18:49] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d8858f20) ICE add candidate: 172.21.38.222:14378, 2130706431
[Sep 27 11:18:49] DEBUG[21655] netsock2.c: Splitting '172.17.0.1' into...
[Sep 27 11:18:49] DEBUG[21655] netsock2.c: ...host '172.17.0.1' and port ''.
[Sep 27 11:18:49] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d8858f20) ICE add candidate: 172.17.0.1:14378, 2130706431
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: RTP instance '0x7f02d8858f20' is setup and ready to go
[Sep 27 11:18:49] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d8858f20) ICE stopped
[Sep 27 11:18:49] DEBUG[21655] netsock2.c: Splitting 'SRVLARGO' into...
[Sep 27 11:18:49] DEBUG[21655] netsock2.c: ...host 'SRVLARGO' and port ''.
[Sep 27 11:18:49] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d8858f20) RTCP setup on RTP instance
[Sep 27 11:18:49] DEBUG[21655] res_srtp.c: local_key64 HDo0imQWTFk+TYRz2b93TXcU80vAfEsKv022UO42 len 40
[Sep 27 11:18:49] DEBUG[21655] res_srtp.c: Adding new policy for SSRC 231267634
[Sep 27 11:18:49] DEBUG[21655] res_srtp.c: SRTP policy activated
[Sep 27 11:18:49] DEBUG[21655] res_srtp.c: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:HDo0imQWTFk+TYRz2b93TXcU80vAfEsKv022UO42
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_sdp_rtp.c:  216
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_sdp_rtp.c:  216
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f02db57d150
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f02db57d150
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Setting tx payload type 9 based on m type on 0x7f02db57d150
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Setting tx payload type 18 based on m type on 0x7f02db57d150
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_sdp_rtp.c:  
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session/pjsip_session_caps.c: '216' Caps for incoming audio call with pref 'local' - remote: (ulaw|alaw|g722|g729|opus|speex16|speex) local: (ulaw|alaw|g729|g722) joint: (ulaw|alaw|g729|g722)
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Crossover copying tx to rx payload mapping 0 (0x7f02db4fead8) from 0x7f02db57d150 to 0x7f02db57d150
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Crossover copying tx to rx payload mapping 8 (0x7f02db4fecb8) from 0x7f02db57d150 to 0x7f02db57d150
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Crossover copying tx to rx payload mapping 9 (0x7f02db4fed08) from 0x7f02db57d150 to 0x7f02db57d150
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Crossover copying tx to rx payload mapping 18 (0x7f02db4feee8) from 0x7f02db57d150 to 0x7f02db57d150
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Crossover copying tx to rx payload mapping 96 (0x7f02d905eab8) from 0x7f02db57d150 to 0x7f02db57d150
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Crossover copying tx to rx payload mapping 97 (0x7f02db5ff9f8) from 0x7f02db57d150 to 0x7f02db57d150
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Crossover copying tx to rx payload mapping 98 (0x7f02d8c6c408) from 0x7f02db57d150 to 0x7f02db57d150
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Crossover copying tx to rx payload mapping 100 (0x7f02d8c6c548) from 0x7f02db57d150 to 0x7f02db57d150
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Copying rx payload mapping 0 (0x7f02db4fead8) from 0x7f02db57d150 to 0x7f02d88590f8
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Copying rx payload mapping 8 (0x7f02db4fecb8) from 0x7f02db57d150 to 0x7f02d88590f8
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Copying rx payload mapping 9 (0x7f02db4fed08) from 0x7f02db57d150 to 0x7f02d88590f8
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Copying rx payload mapping 18 (0x7f02db4feee8) from 0x7f02db57d150 to 0x7f02d88590f8
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Copying rx payload mapping 96 (0x7f02d905eab8) from 0x7f02db57d150 to 0x7f02d88590f8
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Copying rx payload mapping 97 (0x7f02db5ff9f8) from 0x7f02db57d150 to 0x7f02d88590f8
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Copying rx payload mapping 98 (0x7f02d8c6c408) from 0x7f02db57d150 to 0x7f02d88590f8
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Copying rx payload mapping 100 (0x7f02d8c6c548) from 0x7f02db57d150 to 0x7f02d88590f8
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Copying tx payload mapping 0 (0x7f02db4fead8) from 0x7f02db57d150 to 0x7f02d88590f8
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Copying tx payload mapping 8 (0x7f02db4fecb8) from 0x7f02db57d150 to 0x7f02d88590f8
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Copying tx payload mapping 9 (0x7f02db4fed08) from 0x7f02db57d150 to 0x7f02d88590f8
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Copying tx payload mapping 18 (0x7f02db4feee8) from 0x7f02db57d150 to 0x7f02d88590f8
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Copying tx payload mapping 96 (0x7f02d905eab8) from 0x7f02db57d150 to 0x7f02d88590f8
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Copying tx payload mapping 97 (0x7f02db5ff9f8) from 0x7f02db57d150 to 0x7f02d88590f8
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Copying tx payload mapping 98 (0x7f02d8c6c408) from 0x7f02db57d150 to 0x7f02d88590f8
[Sep 27 11:18:49] DEBUG[21655] rtp_engine.c: Copying tx payload mapping 100 (0x7f02d8c6c548) from 0x7f02db57d150 to 0x7f02d88590f8
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_sdp_rtp.c:  
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_sdp_rtp.c:  
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216: Media stream 0:audio-0:audio:sendrecv (ulaw|alaw|g729|g722) handled by audio
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216: Done with stream 0:audio-0:audio:sendrecv (ulaw|alaw|g729|g722)
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216: Handled? yes
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216: Processing streams
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216: Processing stream 0:audio-0:audio:sendrecv (ulaw|alaw|g729|g722)
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216 Adding position 0
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  Using existing media_session
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216 Stream: 0:audio-0:audio:sendrecv (ulaw|alaw|g729|g722)
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_sdp_rtp.c:  216 Type: audio 0:audio-0:audio:sendrecv (ulaw|alaw|g729|g722)
[Sep 27 11:18:49] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d8858f20) RTCP ignoring duplicate property
[Sep 27 11:18:49] DEBUG[21655] res_srtp.c: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:HDo0imQWTFk+TYRz2b93TXcU80vAfEsKv022UO42
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_sdp_rtp.c:  RC: 1
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  Had handler
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216: Stream 0:audio-0:audio:sendrecv (ulaw|alaw|g729|g722) added
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216: Done with 0:audio-0:audio:sendrecv (ulaw|alaw|g729|g722)
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216: Adding bundle groups (if available)
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216: Copying connection details
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216: Processing media 0
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216: Media 0 reset
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  216: Method is INVITE
[Sep 27 11:18:49] DEBUG[21655] chan_pjsip.c:  216
[Sep 27 11:18:49] DEBUG[21655] chan_pjsip.c:  216
[Sep 27 11:18:49] DEBUG[21655] channel_internal_api.c:  <initializing>: Formats: (none)
[Sep 27 11:18:49] DEBUG[21655] channel_internal_api.c:  Channel is being initialized or destroyed
[Sep 27 11:18:49] DEBUG[21655] stasis.c: Creating topic. name: channel:1664270329.3, detail: 
[Sep 27 11:18:49] DEBUG[21655] stasis.c: Topic 'channel:1664270329.3': 0x7f02d84a5730 created
[Sep 27 11:18:49] DEBUG[21655] channel.c: Channel 0x7f02db5ab190 'PJSIP/216-00000003' allocated
[Sep 27 11:18:49] DEBUG[21655] chan_pjsip.c:  Topology:  <0:audio-0:audio:sendrecv (ulaw|alaw|g729|g722)> Formats: (ulaw|alaw|g729|g722)
[Sep 27 11:18:49] DEBUG[21655] chan_pjsip.c:  Compatible? yes
[Sep 27 11:18:49] DEBUG[21505] threadpool.c: Increasing threadpool stasis/pool's size by 1
[Sep 27 11:18:49] DEBUG[21655] channel_internal_api.c:  PJSIP/216-00000003: MultistreamFormats: (ulaw|alaw|g729|g722)
[Sep 27 11:18:49] DEBUG[21655] channel_internal_api.c:  Set native formats but not topology
[Sep 27 11:18:49] DEBUG[21655] channel_internal_api.c:  PJSIP/216-00000003:  <0:audio-0:audio:sendrecv (ulaw|alaw|g729|g722)>
[Sep 27 11:18:49] DEBUG[21655] channel_internal_api.c:  Used provided topology
[Sep 27 11:18:49] DEBUG[21655] chan_pjsip.c:  
[Sep 27 11:18:49] DEBUG[21655] chan_pjsip.c:  PJSIP/216-00000003
[Sep 27 11:18:49] DEBUG[21655] chan_pjsip.c:  PJSIP/216-00000003
[Sep 27 11:18:49] DEBUG[21655] chan_pjsip.c: Started PBX on new PJSIP channel PJSIP/216-00000003
[Sep 27 11:18:49] DEBUG[21655] chan_pjsip.c:  RC: 0
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  PJSIP/216-00000003
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  PJSIP/216-00000003
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  Request: sip:webrtc_client@172.21.38.222 Session: PJSIP/216-00000003
[Sep 27 11:18:49] DEBUG[21655] res_pjsip_session.c:  (null session) Handled request INVITE sip:webrtc_client@172.21.38.222 ? yes
[Sep 27 11:18:49] DEBUG[21681][C-00000003] pbx.c: Launching 'Wait'
[Sep 27 11:18:49] VERBOSE[21681][C-00000003] pbx.c: Executing [webrtc_client@parking:1] Wait("PJSIP/216-00000003", "1") in new stack
3 
[Sep 27 11:18:50] DEBUG[21681][C-00000003] pbx.c: Launching 'Dial'
[Sep 27 11:18:50] VERBOSE[21681][C-00000003] pbx.c: Executing [webrtc_client@parking:2] Dial("PJSIP/216-00000003", "PJSIP/webrtc_client") in new stack
[Sep 27 11:18:50] DEBUG[21681][C-00000003] app_dial.c:  PJSIP/216-00000003: Data: PJSIP/webrtc_client
[Sep 27 11:18:50] DEBUG[21681][C-00000003] chan_pjsip.c:  webrtc_client Topology:  <0:audio-0:audio:sendrecv (ulaw|alaw|g729|g722)>
[Sep 27 11:18:50] DEBUG[21655] chan_pjsip.c:  webrtc_client
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  webrtc_client (null) Topology:  <0:audio-0:audio:sendrecv (ulaw|alaw|g729|g722)>
[Sep 27 11:18:50] DEBUG[21655] config.c: extract double from [3.0] in [-inf, inf] gives [3.000000](0)
[Sep 27 11:18:50] DEBUG[21655] config.c: extract uint from [0] in [0, 4294967295] gives [0](0)
[Sep 27 11:18:50] DEBUG[21655] config.c: extract double from [3.000000] in [-inf, inf] gives [3.000000](0)
[Sep 27 11:18:50] DEBUG[21655] config.c: extract uint from [0] in [0, 4294967295] gives [0](0)
[Sep 27 11:18:50] DEBUG[21655] config.c: extract uint from [0] in [0, 86400] gives [0](0)
[Sep 27 11:18:50] DEBUG[21655] chan_pjsip.c:  webrtc_client
[Sep 27 11:18:50] DEBUG[21655] chan_pjsip.c:  
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session/pjsip_session_caps.c: 'webrtc_client' Caps for outgoing audio call with pref 'remote_merge' - remote: (ulaw|alaw|g729|g722) local: (ulaw|alaw|g729|g722) joint: (ulaw|alaw|g729|g722)
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  
[Sep 27 11:18:50] DEBUG[21655] chan_pjsip.c:  
[Sep 27 11:18:50] DEBUG[21681][C-00000003] chan_pjsip.c:  webrtc_client
[Sep 27 11:18:50] DEBUG[21681][C-00000003] channel_internal_api.c:  <initializing>: Formats: (none)
[Sep 27 11:18:50] DEBUG[21681][C-00000003] channel_internal_api.c:  Channel is being initialized or destroyed
[Sep 27 11:18:50] DEBUG[21681][C-00000003] stasis.c: Creating topic. name: channel:1664270330.4, detail: 
[Sep 27 11:18:50] DEBUG[21681][C-00000003] stasis.c: Topic 'channel:1664270330.4': 0x7f02d84a5370 created
[Sep 27 11:18:50] DEBUG[21681][C-00000003] channel.c: Channel 0x7f02d8cfa080 'PJSIP/webrtc_client-00000004' allocated
[Sep 27 11:18:50] DEBUG[21681][C-00000003] chan_pjsip.c:  Topology:  <0:audio-0:audio:sendrecv (ulaw|alaw|g729|g722)> Formats: (ulaw|alaw|g729|g722)
[Sep 27 11:18:50] DEBUG[21681][C-00000003] chan_pjsip.c:  Compatible? yes
[Sep 27 11:18:50] DEBUG[21681][C-00000003] channel_internal_api.c:  PJSIP/webrtc_client-00000004: MultistreamFormats: (ulaw|alaw|g729|g722)
[Sep 27 11:18:50] DEBUG[21681][C-00000003] channel_internal_api.c:  Set native formats but not topology
[Sep 27 11:18:50] DEBUG[21681][C-00000003] channel_internal_api.c:  PJSIP/webrtc_client-00000004:  <0:audio-0:audio:sendrecv (ulaw|alaw|g729|g722)>
[Sep 27 11:18:50] DEBUG[21681][C-00000003] channel_internal_api.c:  Used provided topology
[Sep 27 11:18:50] DEBUG[21681][C-00000003] chan_pjsip.c:  
[Sep 27 11:18:50] DEBUG[21681][C-00000003] chan_pjsip.c:  Channel: PJSIP/webrtc_client-00000004
[Sep 27 11:18:50] DEBUG[21681][C-00000003] chan_pjsip.c:  PJSIP/webrtc_client-00000004 Topology:  <0:audio-0:audio:sendrecv (ulaw|alaw|g729|g722)>
[Sep 27 11:18:50] DEBUG[21681][C-00000003] chan_pjsip.c:  'call' task pushed
[Sep 27 11:18:50] VERBOSE[21681][C-00000003] app_dial.c: Called PJSIP/webrtc_client
[Sep 27 11:18:50] DEBUG[21655] chan_pjsip.c:  PJSIP/webrtc_client-00000004 Topology:  <0:audio-0:audio:sendrecv (ulaw|alaw|g729|g722)>
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004: Processing streams
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004: Processing stream 0:audio-0:audio:sendrecv (ulaw|alaw|g729|g722)
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004 Adding position 0
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  Creating new media session
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  Setting media session as default for audio
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  Done
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004 Stream: 0:audio-0:audio:sendrecv (ulaw|alaw|g729|g722)
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_sdp_rtp.c:  PJSIP/webrtc_client-00000004 Type: audio 0:audio-0:audio:sendrecv (ulaw|alaw|g729|g722)
[Sep 27 11:18:50] DEBUG[21655] netsock2.c: Splitting '0.0.0.0' into...
[Sep 27 11:18:50] DEBUG[21655] netsock2.c: ...host '0.0.0.0' and port ''.
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_sdp_rtp.c: Transport transport-wss bound to 0.0.0.0: Using it for RTP media.
[Sep 27 11:18:50] DEBUG[21655] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f02d885b690'
[Sep 27 11:18:50] DEBUG[21681][C-00000003] app_dial.c:  PJSIP/216-00000003
[Sep 27 11:18:50] DEBUG[21681][C-00000003] channel.c: Channel PJSIP/webrtc_client-00000004 setting read format path: g722 -> g722
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d885b690) RTP allocated port 15114
[Sep 27 11:18:50] DEBUG[21681][C-00000003] channel.c: Channel PJSIP/216-00000003 setting write format path: g722 -> g722
[Sep 27 11:18:50] DEBUG[21681][C-00000003] channel.c: Channel PJSIP/216-00000003 setting read format path: g722 -> g722
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d885b690) ICE creating session 0.0.0.0:15114 (15114)
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d885b690) ICE create
[Sep 27 11:18:50] DEBUG[21681][C-00000003] channel.c: Channel PJSIP/webrtc_client-00000004 setting write format path: g722 -> g722
[Sep 27 11:18:50] DEBUG[21681][C-00000003] chan_pjsip.c:  PJSIP/216-00000003: Indicated Connected line update
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d885b690) ICE add system candidates
[Sep 27 11:18:50] DEBUG[21655] netsock2.c: Splitting '172.21.38.222' into...
[Sep 27 11:18:50] DEBUG[21655] netsock2.c: ...host '172.21.38.222' and port ''.
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d885b690) ICE add candidate: 172.21.38.222:15114, 2130706431
[Sep 27 11:18:50] DEBUG[21655] netsock2.c: Splitting '172.17.0.1' into...
[Sep 27 11:18:50] DEBUG[21655] netsock2.c: ...host '172.17.0.1' and port ''.
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d885b690) ICE add candidate: 172.17.0.1:15114, 2130706431
[Sep 27 11:18:50] DEBUG[21655] rtp_engine.c: RTP instance '0x7f02d885b690' is setup and ready to go
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d885b690) ICE change number of components 2 -> 1
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d885b690) ICE resetting
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c:  (0x7f02d885b690) ICE nevermind, not ready for a reset
[Sep 27 11:18:50] DEBUG[21655] netsock2.c: Splitting 'SRVLARGO' into...
[Sep 27 11:18:50] DEBUG[21655] netsock2.c: ...host 'SRVLARGO' and port ''.
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d885b690) RTCP setup on RTP instance
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d885b690) DTLS RTP setup
[Sep 27 11:18:50] DEBUG[21681][C-00000003] chan_pjsip.c:  PJSIP/216-00000003
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d885b690) DTLS RTCP setup
[Sep 27 11:18:50] DEBUG[21655] res_srtp.c: local_key64 G5yYLofCfnCQKrNKAlzPnQg943vJJDgFwvgHgGEe len 40
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_sdp_rtp.c: Stream msid: 0x7f02d8c50d10 audio ec6498ab-1f97-4943-a39b-6ec0d25674bc bcc9984f-b6b9-4c5f-951a-998a65b3ef6c
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_sdp_rtp.c:  RC: 1
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  Handled
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004: Stream 0:audio-0:audio:sendrecv (ulaw|alaw|g729|g722) added with mid audio-0
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004: Done with 0:audio-0:audio:sendrecv (ulaw|alaw|g729|g722)
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004: Adding bundle groups (if available)
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004: Copying connection details
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004: Processing media 0
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004: Media 0 reset
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004: Method is INVITE
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004
[Sep 27 11:18:50] DEBUG[21655] res_pjsip/pjsip_resolver.c: Performing SIP DNS resolution of target '172.21.38.236'
[Sep 27 11:18:50] DEBUG[21655] res_pjsip/pjsip_resolver.c: Transport type for target '172.21.38.236' is '(null)'
[Sep 27 11:18:50] DEBUG[21655] res_pjsip/pjsip_resolver.c: Target '172.21.38.236' is an IP address, skipping resolution
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004 Event: TSX_STATE  Inv State: DISCONNCTD
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: Function session_inv_on_state_changed called on event TSX_STATE
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The state change pertains to the endpoint 'webrtc_client(PJSIP/webrtc_client-00000004)'
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The inv session still has an invite_tsx (0x7f02db5db0c8)
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: There is no transaction involved in this state change
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The current inv state is DISCONNCTD
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: PJSIP/webrtc_client-00000004: Source of transaction state change is TRANSPORT_ERROR
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004 TSX State: Terminated  Inv State: DISCONNCTD
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: Function session_inv_on_tsx_state_changed called on event TSX_STATE
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The state change pertains to the endpoint 'webrtc_client(PJSIP/webrtc_client-00000004)'
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The inv session does NOT have an invite_tsx
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The UAC INVITE transaction involved in this state change is 0x7f02db5db0c8
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The current transaction state is Terminated
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The transaction state change event is TRANSPORT_ERROR
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The current inv state is DISCONNCTD
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  Disconnected
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  (null session) TSX State: Terminated  Inv State: DISCONNCTD
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  
[Sep 27 11:18:50] DEBUG[21655] chan_pjsip.c:  RC: 0
[Sep 27 11:18:50] DEBUG[21655] chan_pjsip.c:  PJSIP/webrtc_client-00000004
[Sep 27 11:18:50] DEBUG[21655] chan_pjsip.c:  
[Sep 27 11:18:50] DEBUG[21681][C-00000003] channel.c: PJSIP/216-00000003: Dropping redundant connected line update "webrtc" <parking>.
[Sep 27 11:18:50] DEBUG[21681][C-00000003] channel.c: Channel 0x7f02d8cfa080 'PJSIP/webrtc_client-00000004' hanging up.  Refs: 2
[Sep 27 11:18:50] DEBUG[21681][C-00000003] chan_pjsip.c:  PJSIP/webrtc_client-00000004
[Sep 27 11:18:50] DEBUG[21681][C-00000003] chan_pjsip.c:  Cause: 503
[Sep 27 11:18:50] DEBUG[21525] cdr.c: Finalized CDR for PJSIP/216-00000003 - start 1664270329.398984 answer 0.000000 end 1664270330.414431 dur 1.015 bill 1664270330.414 dispo FAILED
[Sep 27 11:18:50] VERBOSE[21681][C-00000003] app_dial.c: Everyone is busy/congested at this time (1:0/1/0)
[Sep 27 11:18:50] DEBUG[21655] chan_pjsip.c:  PJSIP/webrtc_client-00000004
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/webrtc_client-00000004 Response 503
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: webrtc_client: Destroying SIP session
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d885b690) DTLS srtp - stopped timeout timer'
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d885b690) DTLS srtp - stopped timeout timer'
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d885b690) DTLS stop
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d885b690) DTLS srtp - stopped timeout timer'
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d885b690) DTLS srtp - stopped timeout timer'
[Sep 27 11:18:50] DEBUG[21681][C-00000003] app_dial.c:  PJSIP/216-00000003: No outging channels available
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d885b690) ICE RTP transport deallocating
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d885b690) ICE stopped
[Sep 27 11:18:50] DEBUG[21681][C-00000003] app_dial.c: Exiting with DIALSTATUS=CONGESTION.
[Sep 27 11:18:50] DEBUG[21655] rtp_engine.c: Destroyed RTP instance '0x7f02d885b690'
[Sep 27 11:18:50] DEBUG[21681][C-00000003] app_dial.c:  PJSIP/216-00000003: Done
[Sep 27 11:18:50] DEBUG[21655] channel.c: Channel 0x7f02d8cfa080 'PJSIP/webrtc_client-00000004' destroying
[Sep 27 11:18:50] DEBUG[21681][C-00000003] pbx.c: Launching 'Hangup'
[Sep 27 11:18:50] DEBUG[21525] cdr.c: Finalized CDR for PJSIP/webrtc_client-00000004 - start 1664270330.405554 answer 0.000000 end 1664270330.416349 dur 0.010 bill 1664270330.416 dispo FAILED
[Sep 27 11:18:50] DEBUG[21525] cdr.c: CDR for PJSIP/webrtc_client-00000004 is dialed and has no Party B; discarding
[Sep 27 11:18:50] DEBUG[21655] stasis.c: Destroying topic. name: channel:1664270330.4, detail: 
[Sep 27 11:18:50] DEBUG[21655] stasis.c: Topic 'channel:1664270330.4': 0x7f02d84a5370 destroyed
[Sep 27 11:18:50] DEBUG[21655] channel_internal_api.c:  <initializing>: MultistreamFormats: (nothing)
[Sep 27 11:18:50] DEBUG[21655] channel_internal_api.c:  Channel is being initialized or destroyed
[Sep 27 11:18:50] DEBUG[21655] chan_pjsip.c:  
[Sep 27 11:18:50] DEBUG[21517] devicestate.c: No provider found, checking channel drivers for PJSIP - webrtc_client
[Sep 27 11:18:50] DEBUG[21517] devicestate.c: Changing state for PJSIP/webrtc_client - state 1 (Not in use)
[Sep 27 11:18:50] VERBOSE[21681][C-00000003] pbx.c: Executing [webrtc_client@parking:3] Hangup("PJSIP/216-00000003", "") in new stack
[Sep 27 11:18:50] DEBUG[21681][C-00000003] channel.c: Soft-Hanging (0x20) up channel 'PJSIP/216-00000003'
[Sep 27 11:18:50] DEBUG[21681][C-00000003] pbx.c: Spawn extension (parking,webrtc_client,3) exited non-zero on 'PJSIP/216-00000003'
[Sep 27 11:18:50] VERBOSE[21681][C-00000003] pbx.c: Spawn extension (parking, webrtc_client, 3) exited non-zero on 'PJSIP/216-00000003'
[Sep 27 11:18:50] DEBUG[21681][C-00000003] channel.c: Soft-Hanging (0x10) up channel 'PJSIP/216-00000003'
[Sep 27 11:18:50] DEBUG[21681][C-00000003] channel.c: Channel 0x7f02db5ab190 'PJSIP/216-00000003' hanging up.  Refs: 2
[Sep 27 11:18:50] DEBUG[21681][C-00000003] chan_pjsip.c:  PJSIP/216-00000003
[Sep 27 11:18:50] DEBUG[21681][C-00000003] chan_pjsip.c:  Cause: 503
[Sep 27 11:18:50] DEBUG[21655] chan_pjsip.c:  PJSIP/216-00000003
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/216-00000003 Response 503
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/216-00000003: Method is INVITE, Response is 503 Service Unavailable
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/216-00000003
[Sep 27 11:18:50] VERBOSE[21655] res_pjsip_logger.c: <--- Transmitting SIP response (369 bytes) to UDP:172.21.38.236:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 172.21.38.236:5060;rport=5060;received=172.21.38.236;branch=z9hG4bK.1jH3Y1hfB
Call-ID: ~-NoF7d-dr
From: "216" <sip:216@172.21.38.222>;tag=yO1RnUwCh
To: <sip:webrtc_client@172.21.38.222>;tag=aFKykoc3031flZQBDIhefL5wEf59WXT5
CSeq: 21 INVITE
Server: Asterisk PBX 18.2.2
Reason: Q.850;cause=34
Content-Length:  0


[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/216-00000003 Event: TSX_STATE  Inv State: DISCONNCTD
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: Function session_inv_on_state_changed called on event TSX_STATE
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The state change pertains to the endpoint '216(PJSIP/216-00000003)'
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The inv session still has an invite_tsx (0x7f02db5dccc8)
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: There is no transaction involved in this state change
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The current inv state is DISCONNCTD
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: PJSIP/216-00000003: Source of transaction state change is TX_MSG
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  PJSIP/216-00000003 TSX State: Completed  Inv State: DISCONNCTD
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: Function session_inv_on_tsx_state_changed called on event TSX_STATE
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The state change pertains to the endpoint '216(PJSIP/216-00000003)'
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The inv session still has an invite_tsx (0x7f02db5dccc8)
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The UAS INVITE transaction involved in this state change is 0x7f02db5dccc8
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The current transaction state is Completed
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The transaction state change event is TX_MSG
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The current inv state is DISCONNCTD
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  Disconnected
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  (null session) TSX State: Completed  Inv State: DISCONNCTD
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  
[Sep 27 11:18:50] DEBUG[21655] channel.c: Channel 0x7f02db5ab190 'PJSIP/216-00000003' destroying
[Sep 27 11:18:50] DEBUG[21525] cdr.c: Finalized CDR for PJSIP/216-00000003 - start 1664270330.417342 answer 0.000000 end 1664270330.420250 dur 0.002 bill 1664270330.420 dispo FAILED
[Sep 27 11:18:50] DEBUG[21525] cdr.c: Skipping CDR for PJSIP/216-00000003 since we weren't answered
[Sep 27 11:18:50] DEBUG[21655] stasis.c: Destroying topic. name: channel:1664270329.3, detail: 
[Sep 27 11:18:50] DEBUG[21655] stasis.c: Topic 'channel:1664270329.3': 0x7f02d84a5730 destroyed
[Sep 27 11:18:50] DEBUG[21655] channel_internal_api.c:  <initializing>: MultistreamFormats: (nothing)
[Sep 27 11:18:50] DEBUG[21655] channel_internal_api.c:  Channel is being initialized or destroyed
[Sep 27 11:18:50] DEBUG[21655] chan_pjsip.c:  
[Sep 27 11:18:50] DEBUG[21517] devicestate.c: No provider found, checking channel drivers for PJSIP - 216
[Sep 27 11:18:50] DEBUG[21655] chan_pjsip.c:  216
[Sep 27 11:18:50] DEBUG[21655] chan_pjsip.c:  No channel
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: 216: Destroying SIP session
[Sep 27 11:18:50] DEBUG[21517] devicestate.c: Changing state for PJSIP/216 - state 1 (Not in use)
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d8858f20) DTLS srtp - stopped timeout timer'
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d8858f20) DTLS srtp - stopped timeout timer'
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d8858f20) DTLS stop
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d8858f20) DTLS srtp - stopped timeout timer'
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d8858f20) DTLS srtp - stopped timeout timer'
[Sep 27 11:18:50] DEBUG[21655] res_rtp_asterisk.c: (0x7f02d8858f20) ICE RTP transport deallocating
[Sep 27 11:18:50] DEBUG[21655] rtp_engine.c: Destroyed RTP instance '0x7f02d8858f20'
[Sep 27 11:18:50] VERBOSE[21534] res_pjsip_logger.c: <--- Received SIP request (420 bytes) from UDP:172.21.38.236:5060 --->
ACK sip:webrtc_client@172.21.38.222 SIP/2.0
Via: SIP/2.0/UDP 172.21.38.236:5060;branch=z9hG4bK.1jH3Y1hfB;rport
Call-ID: ~-NoF7d-dr
From: "216" <sip:216@172.21.38.222>;tag=yO1RnUwCh
To: <sip:webrtc_client@172.21.38.222>;tag=aFKykoc3031flZQBDIhefL5wEf59WXT5
Contact: <sip:216@172.21.38.236;transport=udp>;expires=3599;+sip.instance="<urn:uuid:e2c6096f-11a0-0061-a97d-e2c741d445dd>"
Max-Forwards: 70
CSeq: 21 ACK


[Sep 27 11:18:50] DEBUG[21534] res_pjsip/pjsip_distributor.c: Searching for serializer associated with dialog dlg0x7f02d8751a48 for Request msg ACK/cseq=21 (rdata0x7f02db5fd0b8)
[Sep 27 11:18:50] DEBUG[21534] res_pjsip/pjsip_distributor.c: Calculated serializer pjsip/distributor-00000039 to use for Request msg ACK/cseq=21 (rdata0x7f02db5fd0b8)
[Sep 27 11:18:50] DEBUG[21655] netsock2.c: Splitting '172.21.38.236' into...
[Sep 27 11:18:50] DEBUG[21655] netsock2.c: ...host '172.21.38.236' and port ''.
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_endpoint_identifier_ip.c: No identify sections to match against
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_endpoint_identifier_user.c: Attempting identify by From username '216' domain '172.21.38.222'
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_endpoint_identifier_user.c: Identified by From username '216' domain '172.21.38.222'
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  (null session) TSX State: Confirmed  Inv State: DISCONNCTD
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: Function session_inv_on_tsx_state_changed called on event TSX_STATE
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: inv_session 0x7f02db5f7e08 has no ast session
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The inv session still has an invite_tsx (0x7f02db5dccc8)
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The UAS INVITE transaction involved in this state change is 0x7f02db5dccc8
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The current transaction state is Confirmed
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The transaction state change event is RX_MSG
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c: The current inv state is DISCONNCTD
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  Session ended
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:  (null session) TSX State: Confirmed  Inv State: DISCONNCTD
[Sep 27 11:18:50] DEBUG[21655] res_pjsip_session.c:
4

First you’ll want to remove the transport explicitly set on the endpoint. Your old version of Asterisk and PJSIP may not work properly with that. You’ll also want to use a current release of Asterisk, as potentially chasing old issues is never good. Eliminating Docker as a variable to start out with is also generally a good idea.

5

Thanks, by upgrading from Asterisk 18.2.2 to Asterisk 19.5.0 and taking it out of Docker the calls can also be made in the Linphone → SIPML5 direction.
However a problem persists, in this direction only, we have no sound during the call.
Do you have an idea?

Here are the traces (we use the same configuration file as shown above):

[Sep 28 13:04:18] VERBOSE[3353] res_pjsip_logger.c: <--- Received SIP request (1917 bytes) from UDP:172.21.38.236:5060 --->
INVITE sip:webrtc_client@172.21.38.220 SIP/2.0
Via: SIP/2.0/UDP 172.21.38.236:5060;branch=z9hG4bK.DMiY44vzu;rport
From: "216" <sip:216@172.21.38.220>;tag=~Bi9STauS
To: sip:webrtc_client@172.21.38.220
CSeq: 20 INVITE
Call-ID: HTMsYbYSy4
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 1249
Contact: <sip:216@172.21.38.236;transport=udp>;expires=3599;+sip.instance="<urn:uuid:e2c6096f-11a0-0061-a97d-e2c741d445dd>"
User-Agent: Linphone Desktop/4.2.5 (Windows 10 Version 1607, Qt 5.14.2) LinphoneCore/4.4.19

v=0
o=216 1198 1430 IN IP4 172.21.38.236
s=Talk
c=IN IP4 172.21.38.236
t=0 0
a=ice-pwd:cc7ce216b81d04effa90331a
a=ice-ufrag:3f2716ab
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/SAVPF 96 97 98 0 8 9 18 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:66MmKDE2xf7/arnVrdtyneVhzz6jFRUkltm4FR75
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:wgese2oqUn+F4hARMZaM39Outy32WgPnG9HvFAX5
a=crypto:3 AES_256_CM_HMAC_SHA1_80 inline:ygb24lDZCLMAXKk17nlu6C4gQyMLbHR3v8kut4j+33hvgWro+K5OZXLJDKFK4g==
a=crypto:4 AES_256_CM_HMAC_SHA1_32 inline:4OOy0YGXQJyE2TOlXAFHwQ3k8wyB4Evvg5z4OfLw6g2cXgFsjq4zf4IKEaqmwg==
a=candidate:1 1 UDP 2130706303 172.21.38.236 7078 typ host
a=candidate:1 2 UDP 2130706302 172.21.38.236 7079 typ host
a=candidate:2 1 UDP 2130706431 2001:0:2851:782c:38cd:742:53ea:d913 7078 typ host
a=candidate:2 2 UDP 2130706430 2001:0:2851:782c:38cd:742:53ea:d913 7079 typ host
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr

[Sep 28 13:04:18] VERBOSE[31500] res_pjsip_logger.c: <--- Transmitting SIP response (473 bytes) to UDP:172.21.38.236:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.21.38.236:5060;rport=5060;received=172.21.38.236;branch=z9hG4bK.DMiY44vzu
Call-ID: HTMsYbYSy4
From: "216" <sip:216@172.21.38.220>;tag=~Bi9STauS
To: <sip:webrtc_client@172.21.38.220>;tag=z9hG4bK.DMiY44vzu
CSeq: 20 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1664370258/fdc3a1f00bb54ffcb539c6932865697d",opaque="31e60ac525db2943",algorithm=MD5,qop="auth"
Server: FPBX-16.0.21.9(13.38.3)
Content-Length:  0


[Sep 28 13:04:18] VERBOSE[3353] res_pjsip_logger.c: <--- Received SIP request (405 bytes) from UDP:172.21.38.236:5060 --->
ACK sip:webrtc_client@172.21.38.220 SIP/2.0
Via: SIP/2.0/UDP 172.21.38.236:5060;branch=z9hG4bK.DMiY44vzu;rport
Call-ID: HTMsYbYSy4
From: "216" <sip:216@172.21.38.220>;tag=~Bi9STauS
To: <sip:webrtc_client@172.21.38.220>;tag=z9hG4bK.DMiY44vzu
Contact: <sip:216@172.21.38.236;transport=udp>;expires=3599;+sip.instance="<urn:uuid:e2c6096f-11a0-0061-a97d-e2c741d445dd>"
Max-Forwards: 70
CSeq: 20 ACK


[Sep 28 13:04:18] VERBOSE[3353] res_pjsip_logger.c: <--- Received SIP request (2204 bytes) from UDP:172.21.38.236:5060 --->
INVITE sip:webrtc_client@172.21.38.220 SIP/2.0
Via: SIP/2.0/UDP 172.21.38.236:5060;branch=z9hG4bK.AAKvtnfOv;rport
From: "216" <sip:216@172.21.38.220>;tag=~Bi9STauS
To: sip:webrtc_client@172.21.38.220
CSeq: 21 INVITE
Call-ID: HTMsYbYSy4
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 1249
Contact: <sip:216@172.21.38.236;transport=udp>;expires=3599;+sip.instance="<urn:uuid:e2c6096f-11a0-0061-a97d-e2c741d445dd>"
User-Agent: Linphone Desktop/4.2.5 (Windows 10 Version 1607, Qt 5.14.2) LinphoneCore/4.4.19
Authorization:  Digest realm="asterisk", nonce="1664370258/fdc3a1f00bb54ffcb539c6932865697d", algorithm=MD5, opaque="31e60ac525db2943", username="216",  uri="sip:webrtc_client@172.21.38.220", response="de56fcf21ee9468ece90eff3f4a9c040", cnonce="6t0lICEK7MEsulUI", nc=00000001, qop=auth

v=0
o=216 1198 1430 IN IP4 172.21.38.236
s=Talk
c=IN IP4 172.21.38.236
t=0 0
a=ice-pwd:cc7ce216b81d04effa90331a
a=ice-ufrag:3f2716ab
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/SAVPF 96 97 98 0 8 9 18 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:66MmKDE2xf7/arnVrdtyneVhzz6jFRUkltm4FR75
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:wgese2oqUn+F4hARMZaM39Outy32WgPnG9HvFAX5
a=crypto:3 AES_256_CM_HMAC_SHA1_80 inline:ygb24lDZCLMAXKk17nlu6C4gQyMLbHR3v8kut4j+33hvgWro+K5OZXLJDKFK4g==
a=crypto:4 AES_256_CM_HMAC_SHA1_32 inline:4OOy0YGXQJyE2TOlXAFHwQ3k8wyB4Evvg5z4OfLw6g2cXgFsjq4zf4IKEaqmwg==
a=candidate:1 1 UDP 2130706303 172.21.38.236 7078 typ host
a=candidate:1 2 UDP 2130706302 172.21.38.236 7079 typ host
a=candidate:2 1 UDP 2130706431 2001:0:2851:782c:38cd:742:53ea:d913 7078 typ host
a=candidate:2 2 UDP 2130706430 2001:0:2851:782c:38cd:742:53ea:d913 7079 typ host
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr

[Sep 28 13:04:18] VERBOSE[31500] res_pjsip_logger.c: <--- Transmitting SIP response (299 bytes) to UDP:172.21.38.236:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.21.38.236:5060;rport=5060;received=172.21.38.236;branch=z9hG4bK.AAKvtnfOv
Call-ID: HTMsYbYSy4
From: "216" <sip:216@172.21.38.220>;tag=~Bi9STauS
To: <sip:webrtc_client@172.21.38.220>
CSeq: 21 INVITE
Server: FPBX-16.0.21.9(13.38.3)
Content-Length:  0


[Sep 28 13:04:18] VERBOSE[1461][C-0000000c] pbx.c: Executing [webrtc_client@parking:1] Wait("PJSIP/216-00000015", "1") in new stack
[Sep 28 13:04:19] VERBOSE[1461][C-0000000c] pbx.c: Executing [webrtc_client@parking:2] Answer("PJSIP/216-00000015", "") in new stack
[Sep 28 13:04:19] VERBOSE[31500] res_rtp_asterisk.c: 0x7f1070029730 -- Strict RTP learning after remote address set to: 172.21.38.236:7078
[Sep 28 13:04:19] VERBOSE[31500] res_pjsip_logger.c: <--- Transmitting SIP response (972 bytes) to UDP:172.21.38.236:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.21.38.236:5060;rport=5060;received=172.21.38.236;branch=z9hG4bK.AAKvtnfOv
Call-ID: HTMsYbYSy4
From: "216" <sip:216@172.21.38.220>;tag=~Bi9STauS
To: <sip:webrtc_client@172.21.38.220>;tag=228952a3-4049-4bde-8209-7b24dc775fc7
CSeq: 21 INVITE
Server: FPBX-16.0.21.9(13.38.3)
Contact: <sip:172.21.38.220:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   408

v=0
o=- 1198 1432 IN IP4 172.21.38.220
s=Asterisk
c=IN IP4 172.21.38.220
t=0 0
m=audio 13440 RTP/SAVPF 0 8 18 9 100
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:0zmqnA/mpLzkfcwriiStfXiLm/4+ZzGZnE/2Vk94
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[Sep 28 13:04:19] VERBOSE[3353] res_pjsip_logger.c: <--- Received SIP request (670 bytes) from UDP:172.21.38.236:5060 --->
ACK sip:172.21.38.220:5060 SIP/2.0
Via: SIP/2.0/UDP 172.21.38.236:5060;rport;branch=z9hG4bK.-elK~MDak
From: "216" <sip:216@172.21.38.220>;tag=~Bi9STauS
To: <sip:webrtc_client@172.21.38.220>;tag=228952a3-4049-4bde-8209-7b24dc775fc7
CSeq: 21 ACK
Call-ID: HTMsYbYSy4
Max-Forwards: 70
Authorization:  Digest realm="asterisk", nonce="1664370258/fdc3a1f00bb54ffcb539c6932865697d", algorithm=MD5, opaque="31e60ac525db2943", username="216",  uri="sip:webrtc_client@172.21.38.220", response="de56fcf21ee9468ece90eff3f4a9c040", cnonce="6t0lICEK7MEsulUI", nc=00000001, qop=auth
User-Agent: Linphone Desktop/4.2.5 (Windows 10 Version 1607, Qt 5.14.2) LinphoneCore/4.4.19


[Sep 28 13:04:19] VERBOSE[1461][C-0000000c] res_rtp_asterisk.c: 0x7f1070029730 -- Strict RTP switching to RTP target address 172.21.38.236:7078 as source
[Sep 28 13:04:19] VERBOSE[1461][C-0000000c] pbx.c: Executing [webrtc_client@parking:3] Dial("PJSIP/216-00000015", "PJSIP/webrtc_client") in new stack
[Sep 28 13:04:19] VERBOSE[1461][C-0000000c] app_dial.c: Called PJSIP/webrtc_client
[Sep 28 13:04:19] VERBOSE[1461][C-0000000c] app_dial.c: PJSIP/216-00000015 requested media update control 26, passing it to PJSIP/webrtc_client-00000016
[Sep 28 13:04:19] VERBOSE[31501] res_rtp_asterisk.c: DTLS ECDH initialized (automatic), faster PFS enabled
[Sep 28 13:04:19] VERBOSE[31501] res_pjsip_logger.c: <--- Transmitting SIP request (1752 bytes) to WSS:172.21.38.216:3814 --->
INVITE sips:webrtc_client@172.21.38.216:3814;transport=ws;rtcweb-breaker=no SIP/2.0
Via: SIP/2.0/WSS 172.21.38.220:8089;rport;branch=z9hG4bKPj6a816542-58c5-4653-b95d-238c8bea2cca;alias
From: "216" <sip:parking@freepbx.sangoma.local>;tag=74bacc50-6211-4edf-a699-860719775157
To: <sips:webrtc_client@172.21.38.216;rtcweb-breaker=no>
Contact: <sips:asterisk@freepbx.sangoma.local:5060;transport=ws>
Call-ID: e2c8e54d-943c-4efc-a561-2788e1740f16
CSeq: 26088 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: FPBX-16.0.21.9(13.38.3)
Content-Type: application/sdp
Content-Length:   969

v=0
o=- 1187514019 1187514019 IN IP4 172.21.38.220
s=Asterisk
c=IN IP4 172.21.38.220
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio-0
m=audio 12614 UDP/TLS/RTP/SAVPF 0 8 18 9 101
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 84:0D:B4:95:1E:BD:C6:D1:D1:1E:1F:D6:6E:D7:07:C8:9E:1C:C0:9E:A2:FD:33:CB:98:D7:1D:42:03:CB:EB:5B
a=ice-ufrag:6afeae3326b3122c33e6708a5feecf46
a=ice-pwd:01791216411b9acb4d19c94c2f3087e7
a=candidate:Hac1526dc 1 UDP 2130706431 172.21.38.220 12614 typ host
a=candidate:Hac152682 1 UDP 2130706431 172.21.38.130 12614 typ host
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp-mux
a=ssrc:615180356 cname:e0e489fd-cccb-4af1-aa0e-66045eab0202
a=msid:9d022060-6834-474c-9e5d-5c1d5426a896 8c3b0651-de8a-4818-bd7b-87846c3e9240
a=rtcp-fb:* transport-cc
a=mid:audio-0

[Sep 28 13:04:19] VERBOSE[31500] res_pjsip_logger.c: <--- Received SIP response (396 bytes) from WSS:172.21.38.216:3814 --->
SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/WSS 172.21.38.220:8089;rport=8089;branch=z9hG4bKPj6a816542-58c5-4653-b95d-238c8bea2cca;alias
From: "216"<sip:parking@freepbx.sangoma.local>;tag=74bacc50-6211-4edf-a699-860719775157
To: <sips:webrtc_client@172.21.38.216;rtcweb-breaker=no>
Call-ID: e2c8e54d-943c-4efc-a561-2788e1740f16
CSeq: 26088 INVITE
Content-Length: 0


[Sep 28 13:04:19] VERBOSE[31500] res_pjsip_logger.c: <--- Received SIP response (535 bytes) from WSS:172.21.38.216:3814 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WSS 172.21.38.220:8089;rport=8089;branch=z9hG4bKPj6a816542-58c5-4653-b95d-238c8bea2cca;alias
From: "216"<sip:parking@freepbx.sangoma.local>;tag=74bacc50-6211-4edf-a699-860719775157
To: <sips:webrtc_client@172.21.38.216;rtcweb-breaker=no>;tag=mXCoZLbegXV3pMfTMOzw
Contact: <sips:webrtc_client@df7jal23ls0d.invalid;transport=wss>
Call-ID: e2c8e54d-943c-4efc-a561-2788e1740f16
CSeq: 26088 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE


[Sep 28 13:04:19] VERBOSE[1461][C-0000000c] app_dial.c: PJSIP/webrtc_client-00000016 is ringing
[Sep 28 13:04:24] VERBOSE[1461][C-0000000c] res_rtp_asterisk.c: 0x7f1070029730 -- Strict RTP learning complete - Locking on source address 172.21.38.236:7078
[Sep 28 13:04:29] VERBOSE[31500] res_pjsip_logger.c: <--- Received SIP request (637 bytes) from WSS:172.21.38.216:3814 --->
BYE sips:asterisk@freepbx.sangoma.local:5060;transport=ws SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKDiwZgxVu9yJO8A2GoSo7hk22BTAC4fny;rport
From: <sips:webrtc_client@172.21.38.216>;tag=mXCoZLbegXV3pMfTMOzw
To: "216"<sip:parking@freepbx.sangoma.local>;tag=74bacc50-6211-4edf-a699-860719775157
Call-ID: e2c8e54d-943c-4efc-a561-2788e1740f16
CSeq: 15583 BYE
Content-Length: 0
Max-Forwards: 70
Accept-Contact: *;+g.oma.sip-im
Accept-Contact: *;language="en,fr"
Accept-Contact: *;+g.oma.sip-im
Accept-Contact: *;language="en,fr"
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom


[Sep 28 13:04:29] VERBOSE[31500] res_pjsip_logger.c: <--- Transmitting SIP response (410 bytes) to WSS:172.21.38.216:3814 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=3814;received=172.21.38.216;branch=z9hG4bKDiwZgxVu9yJO8A2GoSo7hk22BTAC4fny
Call-ID: e2c8e54d-943c-4efc-a561-2788e1740f16
From: <sips:webrtc_client@172.21.38.216>;tag=mXCoZLbegXV3pMfTMOzw
To: "216" <sip:parking@freepbx.sangoma.local>;tag=74bacc50-6211-4edf-a699-860719775157
CSeq: 15583 BYE
Server: FPBX-16.0.21.9(13.38.3)
Content-Length:  0


[Sep 28 13:04:29] VERBOSE[1461][C-0000000c] app_dial.c: No one is available to answer at this time (1:0/0/0)
[Sep 28 13:04:29] VERBOSE[1461][C-0000000c] pbx.c: Executing [webrtc_client@parking:4] Hangup("PJSIP/216-00000015", "") in new stack
[Sep 28 13:04:29] VERBOSE[1461][C-0000000c] pbx.c: Spawn extension (parking, webrtc_client, 4) exited non-zero on 'PJSIP/216-00000015'
[Sep 28 13:04:29] VERBOSE[31501] res_pjsip_logger.c: <--- Transmitting SIP request (414 bytes) to UDP:172.21.38.236:5060 --->
BYE sip:216@172.21.38.236;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.21.38.220:5060;rport;branch=z9hG4bKPj4a3d3e6f-1dbc-4ad5-b681-612331e61ca2
From: <sip:webrtc_client@172.21.38.220>;tag=228952a3-4049-4bde-8209-7b24dc775fc7
To: "216" <sip:216@172.21.38.220>;tag=~Bi9STauS
Call-ID: HTMsYbYSy4
CSeq: 4286 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: FPBX-16.0.21.9(13.38.3)
Content-Length:  0


[Sep 28 13:04:29] VERBOSE[3353] res_pjsip_logger.c: <--- Received SIP response (412 bytes) from UDP:172.21.38.236:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 172.21.38.220:5060;rport;branch=z9hG4bKPj4a3d3e6f-1dbc-4ad5-b681-612331e61ca2
From: <sip:webrtc_client@172.21.38.220>;tag=228952a3-4049-4bde-8209-7b24dc775fc7
To: "216" <sip:216@172.21.38.220>;tag=~Bi9STauS
Call-ID: HTMsYbYSy4
CSeq: 4286 BYE
User-Agent: Linphone Desktop/4.2.5 (Windows 10 Version 1607, Qt 5.14.2) LinphoneCore/4.4.19
Supported: replaces, outbound, gruu
6

Yes, examine the packet trace to see where media is going. RTP traffic can be displayed using “rtp set debug on”.

7

Finally I can hear sound but only on the side of my Linphone (172.21.38.70:16384), on the side of my Web client (172.21.38.236:54386) no sound.

Here is the RTP trace:

Got  RTP packet from    172.21.38.236:54386 (type 00, seq 023191, ts 2995376244, len 000170)
Sent RTP packet to      172.21.38.70:16384 (type 00, seq 043223, ts 2995376240, len 000160)
Got  RTP packet from    172.21.38.70:16384 (type 00, seq 033023, ts 2991171, len 000160)
Sent RTP packet to      172.21.38.236:54386 (via ICE) (type 00, seq 046833, ts 2991168, len 000170)
Got  RTP packet from    172.21.38.70:16384 (type 00, seq 033024, ts 2991331, len 000160)
Sent RTP packet to      172.21.38.236:54386 (via ICE) (type 00, seq 046834, ts 2991328, len 000170)
Got  RTP packet from    172.21.38.236:54386 (type 00, seq 023192, ts 2995376404, len 000170)
Sent RTP packet to      172.21.38.70:16384 (type 00, seq 043224, ts 2995376400, len 000160)
Got  RTP packet from    172.21.38.236:54386 (type 00, seq 023193, ts 2995376564, len 000170)
Sent RTP packet to      172.21.38.70:16384 (type 00, seq 043225, ts 2995376560, len 000160)
Got  RTP packet from    172.21.38.70:16384 (type 00, seq 033025, ts 2991491, len 000160)
Sent RTP packet to      172.21.38.236:54386 (via ICE) (type 00, seq 046835, ts 2991488, len 000170)
Got  RTP packet from    172.21.38.236:54386 (type 00, seq 023194, ts 2995376724, len 000170)
Sent RTP packet to      172.21.38.70:16384 (type 00, seq 043226, ts 2995376720, len 000160)
Got  RTP packet from    172.21.38.70:16384 (type 00, seq 033026, ts 2991651, len 000160)
Sent RTP packet to      172.21.38.236:54386 (via ICE) (type 00, seq 046836, ts 2991648, len 000170)
Got  RTP packet from    172.21.38.236:54386 (type 00, seq 023195, ts 2995376884, len 000170)
Sent RTP packet to      172.21.38.70:16384 (type 00, seq 043227, ts 2995376880, len 000160)
Got  RTP packet from    172.21.38.70:16384 (type 00, seq 033027, ts 2991811, len 000160)
Sent RTP packet to      172.21.38.236:54386 (via ICE) (type 00, seq 046837, ts 2991808, len 000170)
Got  RTP packet from    172.21.38.70:16384 (type 00, seq 033028, ts 2991971, len 000160)
Sent RTP packet to      172.21.38.236:54386 (via ICE) (type 00, seq 046838, ts 2991968, len 000170)
Got  RTP packet from    172.21.38.236:54386 (type 00, seq 023196, ts 2995377044, len 000170)
Sent RTP packet to      172.21.38.70:16384 (type 00, seq 043228, ts 2995377040, len 000160)
Got  RTP packet from    172.21.38.70:16384 (type 00, seq 033029, ts 2992131, len 000160)
Sent RTP packet to      172.21.38.236:54386 (via ICE) (type 00, seq 046839, ts 2992128, len 000170)
Got  RTP packet from    172.21.38.236:54386 (type 00, seq 023197, ts 2995377204, len 000170)
Sent RTP packet to      172.21.38.70:16384 (type 00, seq 043229, ts 2995377200, len 000160)
Got  RTP packet from    172.21.38.70:16384 (type 00, seq 033030, ts 2992291, len 000160)
Sent RTP packet to      172.21.38.236:54386 (via ICE) (type 00, seq 046840, ts 2992288, len 000170)
Got  RTP packet from    172.21.38.236:54386 (type 00, seq 023198, ts 2995377364, len 000170)
Sent RTP packet to      172.21.38.70:16384 (type 00, seq 043230, ts 2995377360, len 000160)
Got  RTP packet from    172.21.38.70:16384 (type 00, seq 033031, ts 2992451, len 000160)
Sent RTP packet to      172.21.38.236:54386 (via ICE) (type 00, seq 046841, ts 2992448, len 000170)
Got  RTP packet from    172.21.38.236:54386 (type 00, seq 023199, ts 2995377524, len 000170)
Sent RTP packet to      172.21.38.70:16384 (type 00, seq 043231, ts 2995377520, len 000160)
Got  RTP packet from    172.21.38.236:54386 (type 00, seq 023200, ts 2995377684, len 000170)
Sent RTP packet to      172.21.38.70:16384 (type 00, seq 043232, ts 2995377680, len 000160)
Got  RTP packet from    172.21.38.70:16384 (type 00, seq 033032, ts 2992611, len 000160)
Sent RTP packet to      172.21.38.236:54386 (via ICE) (type 00, seq 046842, ts 2992608, len 000170)
Got  RTP packet from    172.21.38.236:54386 (type 00, seq 023201, ts 2995377844, len 000170)
Sent RTP packet to      172.21.38.70:16384 (type 00, seq 043233, ts 2995377840, len 000160)
Got  RTP packet from    172.21.38.70:16384 (type 00, seq 033033, ts 2992771, len 000160)
Sent RTP packet to      172.21.38.236:54386 (via ICE) (type 00, seq 046843, ts 2992768, len 000170)
Got  RTP packet from    172.21.38.70:16384 (type 00, seq 033034, ts 2992931, len 000160)
Sent RTP packet to      172.21.38.236:54386 (via ICE) (type 00, seq 046844, ts 2992928, len 000170)
Got  RTP packet from    172.21.38.236:54386 (type 00, seq 023202, ts 2995378004, len 000170)
Sent RTP packet to      172.21.38.70:16384 (type 00, seq 043234, ts 2995378000, len 000160)
Got  RTP packet from    172.21.38.70:16384 (type 00, seq 033035, ts 2993091, len 000160)
Got  RTP packet from    172.21.38.236:54386 (type 00, seq 023203, ts 2995378164, len 000170)
Sent RTP packet to      172.21.38.70:16384 (type 00, seq 043235, ts 2995378160, len 000160)
Sent RTP packet to      172.21.38.236:54386 (via ICE) (type 00, seq 046845, ts 2993088, len 000170)
Got  RTP packet from    172.21.38.236:54386 (type 00, seq 023204, ts 2995378324, len 000170)
Sent RTP packet to      172.21.38.70:16384 (type 00, seq 043236, ts 2995378320, len 000160)
Got  RTP packet from    172.21.38.70:16384 (type 00, seq 033036, ts 2993251, len 000160)
Sent RTP packet to      172.21.38.236:54386 (via ICE) (type 00, seq 046846, ts 2993248, len 000170)
Got  RTP packet from    172.21.38.236:54386 (type 00, seq 023205, ts 2995378484, len 000170)
Sent RTP packet to      172.21.38.70:16384 (type 00, seq 043237, ts 2995378480, len 000160)
Got  RTP packet from    172.21.38.70:16384 (type 00, seq 033037, ts 2993411, len 000160)
Sent RTP packet to      172.21.38.236:54386 (via ICE) (type 00, seq 046847, ts 2993408, len 000170)
Got  RTP packet from    172.21.38.70:16384 (type 00, seq 033038, ts 2993571, len 000160)
Sent RTP packet to      172.21.38.236:54386 (via ICE) (type 00, seq 046848, ts 2993568, len 000170)
Got  RTP packet from    172.21.38.236:54386 (type 00, seq 023206, ts 2995378644, len 000170)
Sent RTP packet to      172.21.38.70:16384 (type 00, seq 043238, ts 2995378640, len 000160)
Got  RTP packet from    172.21.38.70:16384 (type 00, seq 033039, ts 2993731, len 000160)
Sent RTP packet to      172.21.38.236:54386 (via ICE) (type 00, seq 046849, ts 2993728, len 000170)
Got  RTP packet from    172.21.38.236:54386 (type 00, seq 023207, ts 2995378804, len 000170)
Sent RTP packet to      172.21.38.70:16384 (type 00, seq 043239, ts 2995378800, len 000160)
Got  RTP packet from    172.21.38.70:16384 (type 00, seq 033040, ts 2993891, len 000160)
Sent RTP packet to      172.21.38.236:54386 (via ICE) (type 00, seq 046850, ts 2993888, len 000170)
Got  RTP packet from    172.21.38.236:54386 (type 00, seq 023208, ts 2995378964, len 000170)
Sent RTP packet to      172.21.38.70:16384 (type 00, seq 043240, ts 2995378960, len 000160)
Got  RTP packet from    172.21.38.70:16384 (type 00, seq 033041, ts 2994051, len 000160)
Sent RTP packet to      172.21.38.236:54386 (via ICE) (type 00, seq 046851, ts 2994048, len 000170)
Got  RTP packet from    172.21.38.236:54386 (type 00, seq 023209, ts 2995379124, len 000170)
Sent RTP packet to      172.21.38.70:16384 (type 00, seq 043241, ts 2995379120, len 000160)
Got  RTP packet from    172.21.38.236:54386 (type 00, seq 023210, ts 2995379284, len 000170)
Sent RTP packet to      172.21.38.70:16384 (type 00, seq 043242, ts 2995379280, len 000160)
Got  RTP packet from    172.21.38.70:16384 (type 00, seq 033042, ts 2994211, len 000160)
Sent RTP packet to      172.21.38.236:54386 (via ICE) (type 00, seq 046852, ts 2994208, len 000170)
Got  RTP packet from    172.21.38.70:16384 (type 00, seq 033043, ts 2994371, len 000160)
Sent RTP packet to      172.21.38.236:54386 (via ICE) (type 00, seq 046853, ts 2994368, len 000170)
Got  RTP packet from    172.21.38.236:54386 (type 00, seq 023211, ts 2995379444, len 000170)
Sent RTP packet to      172.21.38.70:16384 (type 00, seq 043243, ts 2995379440, len 000160)
Got  RTP packet from    172.21.38.70:16384 (type 00, seq 033044, ts 2994531, len 000160)
Sent RTP packet to      172.21.38.236:54386 (via ICE) (type 00, seq 046854, ts 2994528, len 000170)
Got  RTP packet from    172.21.38.236:54386 (type 00, seq 023212, ts 2995379604, len 000170)
Sent RTP packet to      172.21.38.70:16384 (type 00, seq 043244, ts 2995379600, len 000160)
Got  RTP packet from    172.21.38.236:54386 (type 00, seq 023213, ts 2995379764, len 000170)
Sent RTP packet to      172.21.38.70:16384 (type 00, seq 043245, ts 2995379760, len 000160)
Got  RTP packet from    172.21.38.70:16384 (type 00, seq 033045, ts 2994691, len 000160)
Sent RTP packet to      172.21.38.236:54386 (via ICE) (type 00, seq 046855, ts 2994688, len 000170)
Got  RTP packet from    172.21.38.236:54386 (type 00, seq 023214, ts 2995379924, len 000170)
Sent RTP packet to      172.21.38.70:16384 (type 00, seq 043246, ts 2995379920, len 000160)
Got  RTP packet from    172.21.38.70:16384 (type 00, seq 033046, ts 2994851, len 000160)
Sent RTP packet to      172.21.38.236:54386 (via ICE) (type 00, seq 046856, ts 2994848, len 000170)
Got  RTP packet from    172.21.38.236:54386 (type 00, seq 023215, ts 2995380084, len 000170)
Sent RTP packet to      172.21.38.70:16384 (type 00, seq 043247, ts 2995380080, len 000160)
Got  RTP packet from    172.21.38.70:16384 (type 00, seq 033047, ts 2995011, len 000160)
Sent RTP packet to      172.21.38.236:54386 (via ICE) (type 00, seq 046857, ts 2995008, len 000170)
Got  RTP packet from    172.21.38.236:54386 (type 00, seq 023216, ts 2995380244, len 000170)
Sent RTP packet to      172.21.38.70:16384 (type 00, seq 043248, ts 2995380240, len 000160)
Got  RTP packet from    172.21.38.70:16384 (type 00, seq 033048, ts 2995171, len 000160)
Sent RTP packet to      172.21.38.236:54386 (via ICE) (type 00, seq 046858, ts 2995168, len 000170)
Got  RTP packet from    172.21.38.70:16384 (type 00, seq 033049, ts 2995331, len 000160)
Sent RTP packet to      172.21.38.236:54386 (via ICE) (type 00, seq 046859, ts 2995328, len 000170)
Got  RTP packet from    172.21.38.236:54386 (type 00, seq 023217, ts 2995380404, len 000170)
Sent RTP packet to      172.21.38.70:16384 (type 00, seq 043249, ts 2995380400, len 000160)
Got  RTP packet from    172.21.38.70:16384 (type 00, seq 033050, ts 2995491, len 000160)
Sent RTP packet to      172.21.38.236:54386 (via ICE) (type 00, seq 046860, ts 2995488, len 000170)
8

According to the RTP trace media is flowing bidirectionally with it, meaning Asterisk is sending audio to it. From what I can see your problem is outside of Asterisk and with the browser. That’s where you’d now need to investigate. Others who spend more time on that side may have suggestions.

9

I’m not sure why you have a packets type of 00. They may not be agreeing on a codec, or transcoding isn’t working.

10

A payload type of 0 is ulaw: RFC 3551 - RTP Profile for Audio and Video Conferences with Minimal Control

11

But Linphone is only offering opus/speex etc, but both sides of the call settle on 00.

12

Statically defined payload types don’t need to have an rtpmap. Multiple codecs were offered, including ulaw:

m=audio 7078 RTP/SAVPF 96 97 98 0 8 9 18 101 99 100
13

Ah, i missed that.

The only thing i can suggest is to use wireshark. Even though the packets are encrypted, they still show up, and if you have packets (inbound and outbound) at the network level, then you should have audio.

14

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