Asterisk with tls on pjsip no audio error

Asterisk Asterisk SIP
1

Hi Guys,

I need your help with an odd problem.
I Have set up TLS on my asterisk.
Register works fine, but when I make a call I get the following error.

[Mar 19 02:15:25] WARNING[1980]: res_pjsip_pubsub.c:3353 pubsub_on_rx_publish_request: No registered publish handler for event presence from 1000
  == Setting global variable 'SIPDOMAIN' to '54.168.134.254'
[Mar 19 02:15:25] ERROR[1980]: res_pjsip_session.c:935 handle_incoming_sdp:  1000: Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing)
[Mar 19 02:15:25] WARNING[1980]: res_pjsip_pubsub.c:3353 pubsub_on_rx_publish_request: No registered publish handler for event presence from 1000

From the debug log I know that it is a problem with codes, I have tried all the codecs, but it is still not working
Anybody that can send me in the right direction. Your help is highly appreciated.

Pjsip set logger on debug.

<--- Received SIP request (2281 bytes) from TLS:219.75.139.45:62477 --->
PUBLISH sip:1000@54.168.134.254 SIP/2.0
Via: SIP/2.0/TLS 192.168.100.87:60769;rport;branch=z9hG4bKPj8346a72c00b14b6ab49eca2440672c73;alias
Max-Forwards: 70
From: "1000" <sip:1000@54.168.134.254>;tag=aa58351e765f4c55b7da1b148ecc2b44
To: "1000" <sip:1000@54.168.134.254>
Call-ID: 6bdb07c7e0fa4c4eaa8c78e5dbedf5c8
CSeq: 1 PUBLISH
Event: presence
Expires: 600
User-Agent: Blink 3.2.0 (Windows)
Content-Type: application/pidf+xml
Content-Length:  1820

<?xml version='1.0' encoding='UTF-8'?>
<presence xmlns:agp-caps="urn:ag-projects:xml:ns:pidf:caps" xmlns:agp-pidf="urn:ag-projects:xml:ns:pidf" xmlns:c="urn:ietf:params:xml:ns:pidf:cipid" xmlns:caps="urn:ietf:params:xml:ns:pidf:caps" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns="urn:ietf:params:xml:ns:pidf" entity="sip%3A1000%4054.168.134.254"><tuple id="SID-e2846047-ecb6-42be-85bf-53371b72b39b"><status><basic>open</basic><agp-pidf:extended>busy</agp-pidf:extended></status><caps:servcaps><caps:audio>true</caps:audio><caps:message>true</caps:message><caps:text>false</caps:text><agp-caps:file-transfer>true</agp-caps:file-transfer><agp-caps:screen-sharing-server>true</agp-caps:screen-sharing-server><agp-caps:screen-sharing-client>true</agp-caps:screen-sharing-client></caps:servcaps><c:display-name>1000</c:display-name><agp-pidf:device-info id="e2846047-ecb6-42be-85bf-53371b72b39b"><agp-pidf:description>wesley</agp-pidf:description><agp-pidf:user-agent>Blink 3.2.0 (Windows)</agp-pidf:user-agent><agp-pidf:time-offset>540</agp-pidf:time-offset></agp-pidf:device-info><rpid:user-input idle-threshold="600">active</rpid:user-input><dm:deviceID>e2846047-ecb6-42be-85bf-53371b72b39b</dm:deviceID><contact>sip%3A1000%4054.168.134.254</contact><note>On the phone</note><timestamp>2021-03-19T11:17:34.096102+09:00</timestamp></tuple><dm:person id="PID-02eb4c078f91b88692a2b9869667d851"><rpid:activities><rpid:busy/></rpid:activities><dm:timestamp>2021-03-19T11:17:34.096102+09:00</dm:timestamp></dm:person><dm:device id="DID-e2846047-ecb6-42be-85bf-53371b72b39b"><dm:deviceID>e2846047-ecb6-42be-85bf-53371b72b39b</dm:deviceID><dm:note>Blink 3.2.0 (Windows) at wesley</dm:note><dm:timestamp>2021-03-19T11:17:34.096102+09:00</dm:timestamp></dm:device></presence>
<--- Transmitting SIP response (573 bytes) to TLS:219.75.139.45:62477 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 192.168.100.87:60769;rport=62477;received=219.75.139.45;branch=z9hG4bKPj8346a72c00b14b6ab49eca2440672c73;alias
Call-ID: 6bdb07c7e0fa4c4eaa8c78e5dbedf5c8
From: "1000" <sip:1000@54.168.134.254>;tag=aa58351e765f4c55b7da1b148ecc2b44
To: "1000" <sip:1000@54.168.134.254>;tag=z9hG4bKPj8346a72c00b14b6ab49eca2440672c73
CSeq: 1 PUBLISH
WWW-Authenticate: Digest realm="asterisk",nonce="1616120253/aa83c4111beb6007b67b1755089845f7",opaque="251fc72563076d9f",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.1.1
Content-Length:  0


<--- Received SIP request (835 bytes) from TLS:219.75.139.45:62477 --->
INVITE sip:1001@54.168.134.254 SIP/2.0
Via: SIP/2.0/TLS 192.168.100.87:60769;rport;branch=z9hG4bKPj80c778fc9f884c21a866fc6fe10cd71f;alias
Max-Forwards: 70
From: "1000" <sip:1000@54.168.134.254>;tag=960cbd206e844ff39cf849acfbacd210
To: <sip:1001@54.168.134.254>
Contact: <sip:72650149@192.168.100.87:57043;transport=tls>
Call-ID: 50d6d495d27141f385248d0120b1398d
CSeq: 20851 INVITE
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER
Supported: replaces, norefersub, gruu
User-Agent: Blink 3.2.0 (Windows)
Content-Type: application/sdp
Content-Length:   240

v=0
o=- 3825141454 3825141454 IN IP4 192.168.100.87
s=Blink 3.2.0 (Windows)
t=0 0
m=audio 50054 RTP/AVP 9 101
c=IN IP4 192.168.100.87
a=rtcp:50055
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

<--- Transmitting SIP response (569 bytes) to TLS:219.75.139.45:62477 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 192.168.100.87:60769;rport=62477;received=219.75.139.45;branch=z9hG4bKPj80c778fc9f884c21a866fc6fe10cd71f;alias
Call-ID: 50d6d495d27141f385248d0120b1398d
From: "1000" <sip:1000@54.168.134.254>;tag=960cbd206e844ff39cf849acfbacd210
To: <sip:1001@54.168.134.254>;tag=z9hG4bKPj80c778fc9f884c21a866fc6fe10cd71f
CSeq: 20851 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1616120253/aa83c4111beb6007b67b1755089845f7",opaque="08babb4e467cd5fa",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.1.1
Content-Length:  0


<--- Received SIP request (2575 bytes) from TLS:219.75.139.45:62477 --->
PUBLISH sip:1000@54.168.134.254 SIP/2.0
Via: SIP/2.0/TLS 192.168.100.87:60769;rport;branch=z9hG4bKPj63057ae9821b4d53a1e5ff91e1899b25;alias
Max-Forwards: 70
From: "1000" <sip:1000@54.168.134.254>;tag=aa58351e765f4c55b7da1b148ecc2b44
To: "1000" <sip:1000@54.168.134.254>
Call-ID: 6bdb07c7e0fa4c4eaa8c78e5dbedf5c8
CSeq: 2 PUBLISH
Event: presence
Expires: 600
User-Agent: Blink 3.2.0 (Windows)
Authorization: Digest username="1000", realm="asterisk", nonce="1616120253/aa83c4111beb6007b67b1755089845f7", uri="sip:1000@54.168.134.254", response="331fb9cfe529880aa10ea465972bcd3b", algorithm=md5, cnonce="32ee2c59249c40d4bdd5262605d6fdea", opaque="251fc72563076d9f", qop=auth, nc=00000001
Content-Type: application/pidf+xml
Content-Length:  1820

<?xml version='1.0' encoding='UTF-8'?>
<presence xmlns:agp-caps="urn:ag-projects:xml:ns:pidf:caps" xmlns:agp-pidf="urn:ag-projects:xml:ns:pidf" xmlns:c="urn:ietf:params:xml:ns:pidf:cipid" xmlns:caps="urn:ietf:params:xml:ns:pidf:caps" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns="urn:ietf:params:xml:ns:pidf" entity="sip%3A1000%4054.168.134.254"><tuple id="SID-e2846047-ecb6-42be-85bf-53371b72b39b"><status><basic>open</basic><agp-pidf:extended>busy</agp-pidf:extended></status><caps:servcaps><caps:audio>true</caps:audio><caps:message>true</caps:message><caps:text>false</caps:text><agp-caps:file-transfer>true</agp-caps:file-transfer><agp-caps:screen-sharing-server>true</agp-caps:screen-sharing-server><agp-caps:screen-sharing-client>true</agp-caps:screen-sharing-client></caps:servcaps><c:display-name>1000</c:display-name><agp-pidf:device-info id="e2846047-ecb6-42be-85bf-53371b72b39b"><agp-pidf:description>wesley</agp-pidf:description><agp-pidf:user-agent>Blink 3.2.0 (Windows)</agp-pidf:user-agent><agp-pidf:time-offset>540</agp-pidf:time-offset></agp-pidf:device-info><rpid:user-input idle-threshold="600">active</rpid:user-input><dm:deviceID>e2846047-ecb6-42be-85bf-53371b72b39b</dm:deviceID><contact>sip%3A1000%4054.168.134.254</contact><note>On the phone</note><timestamp>2021-03-19T11:17:34.096102+09:00</timestamp></tuple><dm:person id="PID-02eb4c078f91b88692a2b9869667d851"><rpid:activities><rpid:busy/></rpid:activities><dm:timestamp>2021-03-19T11:17:34.096102+09:00</dm:timestamp></dm:person><dm:device id="DID-e2846047-ecb6-42be-85bf-53371b72b39b"><dm:deviceID>e2846047-ecb6-42be-85bf-53371b72b39b</dm:deviceID><dm:note>Blink 3.2.0 (Windows) at wesley</dm:note><dm:timestamp>2021-03-19T11:17:34.096102+09:00</dm:timestamp></dm:device></presence>
[Mar 19 02:17:33] WARNING[2135]: res_pjsip_pubsub.c:3353 pubsub_on_rx_publish_request: No registered publish handler for event presence from 1000
<--- Transmitting SIP response (424 bytes) to TLS:219.75.139.45:62477 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/TLS 192.168.100.87:60769;rport=62477;received=219.75.139.45;branch=z9hG4bKPj63057ae9821b4d53a1e5ff91e1899b25;alias
Call-ID: 6bdb07c7e0fa4c4eaa8c78e5dbedf5c8
From: "1000" <sip:1000@54.168.134.254>;tag=aa58351e765f4c55b7da1b148ecc2b44
To: "1000" <sip:1000@54.168.134.254>;tag=z9hG4bKPj63057ae9821b4d53a1e5ff91e1899b25
CSeq: 2 PUBLISH
Server: Asterisk PBX 18.1.1
Content-Length:  0


<--- Received SIP request (426 bytes) from TLS:219.75.139.45:62477 --->
ACK sip:1001@54.168.134.254 SIP/2.0
Via: SIP/2.0/TLS 192.168.100.87:60769;rport;branch=z9hG4bKPj80c778fc9f884c21a866fc6fe10cd71f;alias
Max-Forwards: 70
From: "1000" <sip:1000@54.168.134.254>;tag=960cbd206e844ff39cf849acfbacd210
To: <sip:1001@54.168.134.254>;tag=z9hG4bKPj80c778fc9f884c21a866fc6fe10cd71f
Call-ID: 50d6d495d27141f385248d0120b1398d
CSeq: 20851 ACK
User-Agent: Blink 3.2.0 (Windows)
Content-Length:  0


<--- Received SIP request (1129 bytes) from TLS:219.75.139.45:62477 --->
INVITE sip:1001@54.168.134.254 SIP/2.0
Via: SIP/2.0/TLS 192.168.100.87:60769;rport;branch=z9hG4bKPj50f51411957a4a0fa397e812e4c27749;alias
Max-Forwards: 70
From: "1000" <sip:1000@54.168.134.254>;tag=960cbd206e844ff39cf849acfbacd210
To: <sip:1001@54.168.134.254>
Contact: <sip:72650149@192.168.100.87:57043;transport=tls>
Call-ID: 50d6d495d27141f385248d0120b1398d
CSeq: 20852 INVITE
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER
Supported: replaces, norefersub, gruu
User-Agent: Blink 3.2.0 (Windows)
Authorization: Digest username="1000", realm="asterisk", nonce="1616120253/aa83c4111beb6007b67b1755089845f7", uri="sip:1001@54.168.134.254", response="f543b20d6480aedd56ff74ae7c8f1ebb", algorithm=md5, cnonce="c4748759bbec498c8f2e9de83299354b", opaque="08babb4e467cd5fa", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length:   240

v=0
o=- 3825141454 3825141454 IN IP4 192.168.100.87
s=Blink 3.2.0 (Windows)
t=0 0
m=audio 50054 RTP/AVP 9 101
c=IN IP4 192.168.100.87
a=rtcp:50055
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

  == Setting global variable 'SIPDOMAIN' to '54.168.134.254'
<--- Transmitting SIP response (371 bytes) to TLS:219.75.139.45:62477 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 192.168.100.87:60769;rport=62477;received=219.75.139.45;branch=z9hG4bKPj50f51411957a4a0fa397e812e4c27749;alias
Call-ID: 50d6d495d27141f385248d0120b1398d
From: "1000" <sip:1000@54.168.134.254>;tag=960cbd206e844ff39cf849acfbacd210
To: <sip:1001@54.168.134.254>
CSeq: 20852 INVITE
Server: Asterisk PBX 18.1.1
Content-Length:  0


[Mar 19 02:17:33] ERROR[2135]: res_pjsip_session.c:935 handle_incoming_sdp:  1000: Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing)
<--- Transmitting SIP response (425 bytes) to TLS:219.75.139.45:62477 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/TLS 192.168.100.87:60769;rport=62477;received=219.75.139.45;branch=z9hG4bKPj50f51411957a4a0fa397e812e4c27749;alias
Call-ID: 50d6d495d27141f385248d0120b1398d
From: "1000" <sip:1000@54.168.134.254>;tag=960cbd206e844ff39cf849acfbacd210
To: <sip:1001@54.168.134.254>;tag=dbe7ff3b-714f-4e86-a989-892bc9503ea8
CSeq: 20852 INVITE
Server: Asterisk PBX 18.1.1
Content-Length:  0


<--- Received SIP request (421 bytes) from TLS:219.75.139.45:62477 --->
ACK sip:1001@54.168.134.254 SIP/2.0
Via: SIP/2.0/TLS 192.168.100.87:60769;rport;branch=z9hG4bKPj50f51411957a4a0fa397e812e4c27749;alias
Max-Forwards: 70
From: "1000" <sip:1000@54.168.134.254>;tag=960cbd206e844ff39cf849acfbacd210
To: <sip:1001@54.168.134.254>;tag=dbe7ff3b-714f-4e86-a989-892bc9503ea8
Call-ID: 50d6d495d27141f385248d0120b1398d
CSeq: 20852 ACK
User-Agent: Blink 3.2.0 (Windows)
Content-Length:  0


<--- Received SIP request (2283 bytes) from TLS:219.75.139.45:62477 --->
PUBLISH sip:1000@54.168.134.254 SIP/2.0
Via: SIP/2.0/TLS 192.168.100.87:60769;rport;branch=z9hG4bKPjab78f25168144d428773b5d241293909;alias
Max-Forwards: 70
From: "1000" <sip:1000@54.168.134.254>;tag=77d6a0f59eb04d38b9af04fb08930e9c
To: "1000" <sip:1000@54.168.134.254>
Call-ID: d2dd42e215ec402f9b61a41bc170e3f6
CSeq: 1 PUBLISH
Event: presence
Expires: 600
User-Agent: Blink 3.2.0 (Windows)
Content-Type: application/pidf+xml
Content-Length:  1822

<?xml version='1.0' encoding='UTF-8'?>
<presence xmlns:agp-caps="urn:ag-projects:xml:ns:pidf:caps" xmlns:agp-pidf="urn:ag-projects:xml:ns:pidf" xmlns:c="urn:ietf:params:xml:ns:pidf:cipid" xmlns:caps="urn:ietf:params:xml:ns:pidf:caps" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns="urn:ietf:params:xml:ns:pidf" entity="sip%3A1000%4054.168.134.254"><tuple id="SID-e2846047-ecb6-42be-85bf-53371b72b39b"><status><basic>open</basic><agp-pidf:extended>available</agp-pidf:extended></status><caps:servcaps><caps:audio>true</caps:audio><caps:message>true</caps:message><caps:text>false</caps:text><agp-caps:file-transfer>true</agp-caps:file-transfer><agp-caps:screen-sharing-server>true</agp-caps:screen-sharing-server><agp-caps:screen-sharing-client>true</agp-caps:screen-sharing-client></caps:servcaps><c:display-name>1000</c:display-name><agp-pidf:device-info id="e2846047-ecb6-42be-85bf-53371b72b39b"><agp-pidf:description>wesley</agp-pidf:description><agp-pidf:user-agent>Blink 3.2.0 (Windows)</agp-pidf:user-agent><agp-pidf:time-offset>540</agp-pidf:time-offset></agp-pidf:device-info><rpid:user-input idle-threshold="600">active</rpid:user-input><dm:deviceID>e2846047-ecb6-42be-85bf-53371b72b39b</dm:deviceID><contact>sip%3A1000%4054.168.134.254</contact><timestamp>2021-03-19T11:17:34.341415+09:00</timestamp></tuple><dm:person id="PID-02eb4c078f91b88692a2b9869667d851"><rpid:activities><rpid:other>available</rpid:other></rpid:activities><dm:timestamp>2021-03-19T11:17:34.341415+09:00</dm:timestamp></dm:person><dm:device id="DID-e2846047-ecb6-42be-85bf-53371b72b39b"><dm:deviceID>e2846047-ecb6-42be-85bf-53371b72b39b</dm:deviceID><dm:note>Blink 3.2.0 (Windows) at wesley</dm:note><dm:timestamp>2021-03-19T11:17:34.341415+09:00</dm:timestamp></dm:device></presence>
<--- Transmitting SIP response (573 bytes) to TLS:219.75.139.45:62477 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 192.168.100.87:60769;rport=62477;received=219.75.139.45;branch=z9hG4bKPjab78f25168144d428773b5d241293909;alias
Call-ID: d2dd42e215ec402f9b61a41bc170e3f6
From: "1000" <sip:1000@54.168.134.254>;tag=77d6a0f59eb04d38b9af04fb08930e9c
To: "1000" <sip:1000@54.168.134.254>;tag=z9hG4bKPjab78f25168144d428773b5d241293909
CSeq: 1 PUBLISH
WWW-Authenticate: Digest realm="asterisk",nonce="1616120253/aa83c4111beb6007b67b1755089845f7",opaque="5b0dfcf23621a6f2",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.1.1
Content-Length:  0


<--- Received SIP request (2577 bytes) from TLS:219.75.139.45:62477 --->
PUBLISH sip:1000@54.168.134.254 SIP/2.0
Via: SIP/2.0/TLS 192.168.100.87:60769;rport;branch=z9hG4bKPj55ba09b696f8464d8c6d7ede8db62c64;alias
Max-Forwards: 70
From: "1000" <sip:1000@54.168.134.254>;tag=77d6a0f59eb04d38b9af04fb08930e9c
To: "1000" <sip:1000@54.168.134.254>
Call-ID: d2dd42e215ec402f9b61a41bc170e3f6
CSeq: 2 PUBLISH
Event: presence
Expires: 600
User-Agent: Blink 3.2.0 (Windows)
Authorization: Digest username="1000", realm="asterisk", nonce="1616120253/aa83c4111beb6007b67b1755089845f7", uri="sip:1000@54.168.134.254", response="41ba78d73ffa88ed4b4f9869e8a57980", algorithm=md5, cnonce="3443501673bd4f6b99f0cba429599c3f", opaque="5b0dfcf23621a6f2", qop=auth, nc=00000001
Content-Type: application/pidf+xml
Content-Length:  1822

<?xml version='1.0' encoding='UTF-8'?>
<presence xmlns:agp-caps="urn:ag-projects:xml:ns:pidf:caps" xmlns:agp-pidf="urn:ag-projects:xml:ns:pidf" xmlns:c="urn:ietf:params:xml:ns:pidf:cipid" xmlns:caps="urn:ietf:params:xml:ns:pidf:caps" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns="urn:ietf:params:xml:ns:pidf" entity="sip%3A1000%4054.168.134.254"><tuple id="SID-e2846047-ecb6-42be-85bf-53371b72b39b"><status><basic>open</basic><agp-pidf:extended>available</agp-pidf:extended></status><caps:servcaps><caps:audio>true</caps:audio><caps:message>true</caps:message><caps:text>false</caps:text><agp-caps:file-transfer>true</agp-caps:file-transfer><agp-caps:screen-sharing-server>true</agp-caps:screen-sharing-server><agp-caps:screen-sharing-client>true</agp-caps:screen-sharing-client></caps:servcaps><c:display-name>1000</c:display-name><agp-pidf:device-info id="e2846047-ecb6-42be-85bf-53371b72b39b"><agp-pidf:description>wesley</agp-pidf:description><agp-pidf:user-agent>Blink 3.2.0 (Windows)</agp-pidf:user-agent><agp-pidf:time-offset>540</agp-pidf:time-offset></agp-pidf:device-info><rpid:user-input idle-threshold="600">active</rpid:user-input><dm:deviceID>e2846047-ecb6-42be-85bf-53371b72b39b</dm:deviceID><contact>sip%3A1000%4054.168.134.254</contact><timestamp>2021-03-19T11:17:34.341415+09:00</timestamp></tuple><dm:person id="PID-02eb4c078f91b88692a2b9869667d851"><rpid:activities><rpid:other>available</rpid:other></rpid:activities><dm:timestamp>2021-03-19T11:17:34.341415+09:00</dm:timestamp></dm:person><dm:device id="DID-e2846047-ecb6-42be-85bf-53371b72b39b"><dm:deviceID>e2846047-ecb6-42be-85bf-53371b72b39b</dm:deviceID><dm:note>Blink 3.2.0 (Windows) at wesley</dm:note><dm:timestamp>2021-03-19T11:17:34.341415+09:00</dm:timestamp></dm:device></presence>
[Mar 19 02:17:33] WARNING[2135]: res_pjsip_pubsub.c:3353 pubsub_on_rx_publish_request: No registered publish handler for event presence from 1000
<--- Transmitting SIP response (424 bytes) to TLS:219.75.139.45:62477 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/TLS 192.168.100.87:60769;rport=62477;received=219.75.139.45;branch=z9hG4bKPj55ba09b696f8464d8c6d7ede8db62c64;alias
Call-ID: d2dd42e215ec402f9b61a41bc170e3f6
From: "1000" <sip:1000@54.168.134.254>;tag=77d6a0f59eb04d38b9af04fb08930e9c
To: "1000" <sip:1000@54.168.134.254>;tag=z9hG4bKPj55ba09b696f8464d8c6d7ede8db62c64
CSeq: 2 PUBLISH
Server: Asterisk PBX 18.1.1
Content-Length:  0

My pjsip file.

[transport-tls]
type=transport
protocol=tls
bind=0.0.0.0:5061
cert_file=/etc/asterisk/keys/asterisk.crt
priv_key_file=/etc/asterisk/keys/asterisk.key
method=sslv23

; Sip Accounts Template
[aor-single-reg](!)
type=aor
max_contacts=1
remove_existing=yes
 
[auth-userpass](!)
type=auth
auth_type=userpass

[endpoint-basic](!)
type=endpoint
context=extensions
disallow=all
allow=g722
dtmf_mode=rfc4733
media_encryption=sdes

; Sip Accounts

; 1000
[1000](aor-single-reg)
[1000](auth-userpass)
username=1000
password=1000

[1000](endpoint-basic)
auth=1000
outbound_auth=1000
aors=1000

; 1001
[1001](aor-single-reg)
[1001](auth-userpass)
username=1001
password=1001

[1001](endpoint-basic)
auth=1001
outbound_auth=1001
aors=1001
2

You told your extension / trunk to use encrypted RTP, but the caller doesn’t provide it. See the SDP of the Invite:

This should read “m=audio 50054 RTP/SAVP 9 101”

That’s why asterisk tells “Couldn’t negotiate stream 0:audio-0:audio:sendrecv (nothing)” because there is no valid codec given, which fulfills the requirements defined by you. You should tell the Caller to provide encryption, too or you must disable encryption or you have to change your configuration to allow unencrypted RTP, too (opportunistic SRTP).

3

@micha Thank you for your response.
That points me a bit in the right direction.

I have created the pjsip configuration from the Asterisk website.
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial

So basically the settings of Asterisk are correct, but I need to provide encryption from the client?
How can you allow unencrypted RTP or SRTP is there a command for the pjsip config file.

Thank you for your help.
It’s highly appreciated.

4

For the understanding: Asterisk defines, how the clients have to behave. If you tell asterisk, that a client should use SRTP, you obviously have to tell the client, that it has to provide SRTP, too. Therefore you have firstly to think about, what you want to achieve.

=> Yes

Take a look at the media_encryption switch for the configuration of asterisk.

5

Thank you,
I’m going to take a look at what I did wrong with my pjsip file and try to fix it.

6

If you need media encryption, you have to fix this on the peer, not on Asterisk. Telling Asterisk to accept AVP as well as SAVP will result in the call succeeding, but with no media encryption.

7

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