Need help undertending SRTP log

Asterisk Asterisk Support
1

Hi,

When i cal i get:
WARNING[1551][C-00000013]: chan_sip.c:10427 process_sdp: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio

My sip.conf:

[general]
context=public
allowoverlap=no
;udp=no
udpbindaddr=::
tcpenable=yes
tcpbindaddr=::
srvlookup=yes
qualify=yes
nat=no
dtmfmode=auto
transport=tls
;media_encryption=sdes
host=dynamic
videosupport=no
directmedia=yes
disallow=all
allow=g722
allow=ulaw
allow=alaw

;;TLS;;
tlsenable=yes
tlscertfile=/etc/asterisk/keys/pbx.xxxxxx.pem ; certificaat + key
tlscafile=/etc/asterisk/keys/ca.crt ; RapidSSL_CA_bundle.pem
tlsbindaddr=::
tlsdontverifyserver=no
tlscipher=DES-CBC3-SHA
tlsclientmethod=tlsv1

;;SRTP;;
media_encryption=sdes
encryption=yes

SIP TLS works perfect.
Module srtp.so is loaded

Any ideas are welcome!!

2

The peer tried to use SRTP but did not provide an encryption key in their SDP. You haven’t provided their SDP, so we can’t see the details.

3

What do you mean with the client’s SDP?

4

You are using SIP. When a SIP client tries to establish a session, it sends data using session description protocol. That is what is used to request SRTP. In order to have SRTP, that SDP must specify an encryption key to be used for the media. en.wikipedia.org/wiki/Session_De … n_Protocol

sip set debug on will allow you to see what SDP is being sent.

5

Thanks alot for your reply!!!
Here the sip log:

<— SIP read from TLS:[2001:980:7936:0:204:13ff:fe75:5744]:55578 —>
INVITE sip:101@pbx.picopbx.nl;user=phone SIP/2.0
Via: SIP/2.0/TLS [2001:980:7936:0:204:13ff:fe75:5744]:55578;branch=z9hG4bK-x30rsm8n1hvy;rport
From: “pico two” sip:pico-02@pbx.picopbx.nl;tag=aqd2z6jdke
To: sip:101@pbx.picopbx.nl;user=phone
Call-ID: D279475571D105175DA8E2B4BE846630-pox0zya0tz3r
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: snom715/8.7.5.8.2
Contact: sip:pico-02@[2001:980:7936:0:204:13ff:fe75:5744]:55578;transport=tls;line=a0bxgm4i;reg-id=1
X-Serialnumber: 000413755744
P-Key-Flags: resolution=“31x13”, keys="4"
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 662

v=0
o=root 1085102407 1085102407 IN IP6 2001:980:7936:0:204:13ff:fe75:5744
s=call
c=IN IP6 2001:980:7936:0:204:13ff:fe75:5744
t=0 0
m=audio 57382 RTP/AVP 9 0 8 3 97 98 99 100 112 113 114 115 18 101
a=direction:both
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 G726-16/8000
a=rtpmap:98 G726-24/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:100 G726-40/8000
a=rtpmap:112 AAL2-G726-16/8000
a=rtpmap:113 AAL2-G726-24/8000
a=rtpmap:114 AAL2-G726-32/8000
a=rtpmap:115 AAL2-G726-40/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
— (19 headers 25 lines) —
Sending to [2001:980:7936:0:204:13ff:fe75:5744]:55578 (no NAT)
Sending to [2001:980:7936:0:204:13ff:fe75:5744]:55578 (no NAT)
Using INVITE request as basis request - D279475571D105175DA8E2B4BE846630-pox0zya0tz3r
Found peer ‘pico-02’ for ‘pico-02’ from [2001:980:7936:0:204:13ff:fe75:5744]:55578

<— Reliably Transmitting (no NAT) to [2001:980:7936:0:204:13ff:fe75:5744]:55578 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS [2001:980:7936:0:204:13ff:fe75:5744]:55578;branch=z9hG4bK-x30rsm8n1hvy;received=2001:980:7936:0:204:13ff:fe75:5744;rport=55578
From: “pico two” sip:pico-02@pbx.picopbx.nl;tag=aqd2z6jdke
To: sip:101@pbx.picopbx.nl;user=phone;tag=as38d81c39
Call-ID: D279475571D105175DA8E2B4BE846630-pox0zya0tz3r
CSeq: 1 INVITE
Server: Asterisk PBX 13.3.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="3731b1ab"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘D279475571D105175DA8E2B4BE846630-pox0zya0tz3r’ in 6400 ms (Method: INVITE)

<— SIP read from TLS:[2001:980:7936:0:204:13ff:fe75:5744]:55578 —>
ACK sip:101@pbx.picopbx.nl;user=phone SIP/2.0
Via: SIP/2.0/TLS [2001:980:7936:0:204:13ff:fe75:5744]:55578;branch=z9hG4bK-x30rsm8n1hvy;rport
From: “pico two” sip:pico-02@pbx.picopbx.nl;tag=aqd2z6jdke
To: sip:101@pbx.picopbx.nl;user=phone;tag=as38d81c39
Call-ID: D279475571D105175DA8E2B4BE846630-pox0zya0tz3r
CSeq: 1 ACK
Max-Forwards: 70
User-Agent: snom715/8.7.5.8.2
Contact: sip:pico-02@[2001:980:7936:0:204:13ff:fe75:5744]:55578;transport=tls;line=a0bxgm4i;reg-id=1
Content-Length: 0

<------------->
— (10 headers 0 lines) —

<— SIP read from TLS:[2001:980:7936:0:204:13ff:fe75:5744]:55578 —>
INVITE sip:101@pbx.picopbx.nl;user=phone SIP/2.0
Via: SIP/2.0/TLS [2001:980:7936:0:204:13ff:fe75:5744]:55578;branch=z9hG4bK-hcuabhlpzrc2;rport
From: “pico two” sip:pico-02@pbx.picopbx.nl;tag=aqd2z6jdke
To: sip:101@pbx.picopbx.nl;user=phone
Call-ID: D279475571D105175DA8E2B4BE846630-pox0zya0tz3r
CSeq: 2 INVITE
Max-Forwards: 70
User-Agent: snom715/8.7.5.8.2
Contact: sip:pico-02@[2001:980:7936:0:204:13ff:fe75:5744]:55578;transport=tls;line=a0bxgm4i;reg-id=1
X-Serialnumber: 000413755744
P-Key-Flags: resolution=“31x13”, keys=“4"
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Authorization: Digest username=“pico-02”,realm=“asterisk”,nonce=“3731b1ab”,uri="sip:101@pbx.picopbx.nl;user=phone”,response=“4c8c1a39f57b2369d63c3086165b7b62”,algorithm=MD5
Content-Type: application/sdp
Content-Length: 662

v=0
o=root 1085102407 1085102407 IN IP6 2001:980:7936:0:204:13ff:fe75:5744
s=call
c=IN IP6 2001:980:7936:0:204:13ff:fe75:5744
t=0 0
m=audio 57382 RTP/AVP 9 0 8 3 97 98 99 100 112 113 114 115 18 101
a=direction:both
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 G726-16/8000
a=rtpmap:98 G726-24/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:100 G726-40/8000
a=rtpmap:112 AAL2-G726-16/8000
a=rtpmap:113 AAL2-G726-24/8000
a=rtpmap:114 AAL2-G726-32/8000
a=rtpmap:115 AAL2-G726-40/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
— (20 headers 25 lines) —
Sending to [2001:980:7936:0:204:13ff:fe75:5744]:55578 (no NAT)
Using INVITE request as basis request - D279475571D105175DA8E2B4BE846630-pox0zya0tz3r
Found peer ‘pico-02’ for ‘pico-02’ from [2001:980:7936:0:204:13ff:fe75:5744]:55578
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 99
Found RTP audio format 100
Found RTP audio format 112
Found RTP audio format 113
Found RTP audio format 114
Found RTP audio format 115
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found unknown media description format G726-16 for ID 97
Found unknown media description format G726-24 for ID 98
Found audio description format G726-32 for ID 99
Found unknown media description format G726-40 for ID 100
Found unknown media description format AAL2-G726-16 for ID 112
Found unknown media description format AAL2-G726-24 for ID 113
Found audio description format AAL2-G726-32 for ID 114
Found unknown media description format AAL2-G726-40 for ID 115
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
[May 4 15:53:24] WARNING[1585][C-00000001]: chan_sip.c:10520 process_sdp: Matched device setup to use SRTP, but request was not!

<— Reliably Transmitting (no NAT) to [2001:980:7936:0:204:13ff:fe75:5744]:55578 —>
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/TLS [2001:980:7936:0:204:13ff:fe75:5744]:55578;branch=z9hG4bK-hcuabhlpzrc2;received=2001:980:7936:0:204:13ff:fe75:5744;rport=55578
From: “pico two” sip:pico-02@pbx.picopbx.nl;tag=aqd2z6jdke
To: sip:101@pbx.picopbx.nl;user=phone;tag=as38d81c39
Call-ID: D279475571D105175DA8E2B4BE846630-pox0zya0tz3r
CSeq: 2 INVITE
Server: Asterisk PBX 13.3.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘D279475571D105175DA8E2B4BE846630-pox0zya0tz3r’ in 6400 ms (Method: INVITE)

<— SIP read from TLS:[2001:980:7936:0:204:13ff:fe75:5744]:55578 —>
ACK sip:101@pbx.picopbx.nl;user=phone SIP/2.0
Via: SIP/2.0/TLS [2001:980:7936:0:204:13ff:fe75:5744]:55578;branch=z9hG4bK-hcuabhlpzrc2;rport
From: “pico two” sip:pico-02@pbx.picopbx.nl;tag=aqd2z6jdke
To: sip:101@pbx.picopbx.nl;user=phone;tag=as38d81c39
Call-ID: D279475571D105175DA8E2B4BE846630-pox0zya0tz3r
CSeq: 2 ACK
Max-Forwards: 70
User-Agent: snom715/8.7.5.8.2
Contact: sip:pico-02@[2001:980:7936:0:204:13ff:fe75:5744]:55578;transport=tls;line=a0bxgm4i;reg-id=1
Content-Length: 0

<------------->
— (10 headers 0 lines) —
pbx*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups

6

sip show peer:

  • Name : pico-01
    Description :
    Secret :
    MD5Secret :
    Remote Secret:
    Context : intern
    Record On feature : automon
    Record Off feature : automon
    Subscr.Cont. :
    Language :
    Tonezone :
    AMA flags : Unknown
    Transfer mode: open
    CallingPres : Presentation Allowed, Not Screened
    Callgroup :
    Pickupgroup :
    Named Callgr :
    Nam. Pickupgr:
    MOH Suggest :
    Mailbox :
    VM Extension : asterisk
    LastMsgsSent : 0/0
    Call limit : 0
    Max forwards : 0
    Dynamic : Yes
    Callerid : “” <>
    MaxCallBR : 384 kbps
    Expire : 3413
    Insecure : no
    Force rport : No
    Symmetric RTP: No
    ACL : No
    DirectMedACL : No
    T.38 support : No
    T.38 EC mode : Unknown
    T.38 MaxDtgrm: 4294967295
    DirectMedia : Yes
    PromiscRedir : No
    User=Phone : No
    Video Support: No
    Text Support : No
    Ign SDP ver : No
    Trust RPID : No
    Send RPID : No
    Path support : No
    Path : N/A
    TrustIDOutbnd: Legacy
    Subscriptions: Yes
    Overlap dial : No
    DTMFmode : auto
    Timer T1 : 500
    Timer B : 32000
    ToHost :
    Addr->IP : [2001:980:7936:0:204:13ff:fe75:5722]:45277
    Defaddr->IP : (null)
    Prim.Transp. : TLS
    Allowed.Trsp : TLS
    Def. Username: pico-01
    SIP Options : (none)
    Codecs : (ulaw|alaw|g722)
    Auto-Framing : No
    Status : OK (35 ms)
    Useragent : snom715/8.7.5.8.2
    Reg. Contact : sip:pico-01@[2001:980:7936:0:204:13ff:fe75:5722]:45277;transport=tls;line=l4u12ubd
    Qualify Freq : 60000 ms
    Keepalive : 0 ms
    Sess-Timers : Accept
    Sess-Refresh : uas
    Sess-Expires : 1800 secs
    Min-Sess : 90 secs
    RTP Engine : asterisk
    Parkinglot :
    Use Reason : No
    Encryption : Ye
7

You are insisting on media encryption. The peer isn’t prepared to offer it. As it says:

Either remove the encryption requirement, or enable it in the peer.

8

Thank your for your reply!
I am really trying to understand how srtp works and what it needs, sadly there is verry little documentation on the net.

---------------------------------------log

Really destroying SIP dialog ‘dd235bd9e2c2389ca82f56d7175a4198’ Method: ACK

<— SIP read from TLS:[2001:980:7936:0:204:13ff:fe75:5744]:32789 —>
INVITE sip:101@pbx.picopbx.nl;user=phone SIP/2.0
Via: SIP/2.0/TLS [2001:980:7936:0:204:13ff:fe75:5744]:32789;branch=z9hG4bK-b8h6nuc0ov5m;rport
From: “pico two” sip:pico-02@pbx.picopbx.nl;tag=escuiu6jhn
To: sip:101@pbx.picopbx.nl;user=phone
Call-ID: 47044A553432012313A8628FBE846630-7nev3hg2uc1t
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: snom715/8.7.5.8.2
Contact: sip:pico-02@[2001:980:7936:0:204:13ff:fe75:5744]:32789;transport=tls;line=mjvft1a2;reg-id=1
X-Serialnumber: 000413755744
P-Key-Flags: resolution=“31x13”, keys="4"
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 729

v=0
o=root 1771457770 1771457770 IN IP6 2001:980:7936:0:204:13ff:fe75:5744
s=call
c=IN IP6 2001:980:7936:0:204:13ff:fe75:5744
t=0 0
m=audio 61136 RTP/SAVP 9 0 8 3 97 98 99 100 112 113 114 115 18 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:uOx0mT6szY6Iy2/yfz+Myl12OjzBFbcxLqOL/AS8
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 G726-16/8000
a=rtpmap:98 G726-24/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:100 G726-40/8000
a=rtpmap:112 AAL2-G726-16/8000
a=rtpmap:113 AAL2-G726-24/8000
a=rtpmap:114 AAL2-G726-32/8000
a=rtpmap:115 AAL2-G726-40/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
— (19 headers 25 lines) —
Sending to [2001:980:7936:0:204:13ff:fe75:5744]:32789 (no NAT)
Sending to [2001:980:7936:0:204:13ff:fe75:5744]:32789 (no NAT)
Using INVITE request as basis request - 47044A553432012313A8628FBE846630-7nev3hg2uc1t
Found peer ‘pico-02’ for ‘pico-02’ from [2001:980:7936:0:204:13ff:fe75:5744]:32789

<— Reliably Transmitting (no NAT) to [2001:980:7936:0:204:13ff:fe75:5744]:32789 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS [2001:980:7936:0:204:13ff:fe75:5744]:32789;branch=z9hG4bK-b8h6nuc0ov5m;received=2001:980:7936:0:204:13ff:fe75:5744;rport=32789
From: “pico two” sip:pico-02@pbx.picopbx.nl;tag=escuiu6jhn
To: sip:101@pbx.picopbx.nl;user=phone;tag=as539b87ab
Call-ID: 47044A553432012313A8628FBE846630-7nev3hg2uc1t
CSeq: 1 INVITE
Server: Asterisk PBX 13.3.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="12bbae7b"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘47044A553432012313A8628FBE846630-7nev3hg2uc1t’ in 6400 ms (Method: INVITE)

<— SIP read from TLS:[2001:980:7936:0:204:13ff:fe75:5744]:32789 —>
ACK sip:101@pbx.picopbx.nl;user=phone SIP/2.0
Via: SIP/2.0/TLS [2001:980:7936:0:204:13ff:fe75:5744]:32789;branch=z9hG4bK-b8h6nuc0ov5m;rport
From: “pico two” sip:pico-02@pbx.picopbx.nl;tag=escuiu6jhn
To: sip:101@pbx.picopbx.nl;user=phone;tag=as539b87ab
Call-ID: 47044A553432012313A8628FBE846630-7nev3hg2uc1t
CSeq: 1 ACK
Max-Forwards: 70
User-Agent: snom715/8.7.5.8.2
Contact: sip:pico-02@[2001:980:7936:0:204:13ff:fe75:5744]:32789;transport=tls;line=mjvft1a2;reg-id=1
Content-Length: 0

<------------->
— (10 headers 0 lines) —

<— SIP read from TLS:[2001:980:7936:0:204:13ff:fe75:5744]:32789 —>
INVITE sip:101@pbx.picopbx.nl;user=phone SIP/2.0
Via: SIP/2.0/TLS [2001:980:7936:0:204:13ff:fe75:5744]:32789;branch=z9hG4bK-mw6cgfvuxl6i;rport
From: “pico two” sip:pico-02@pbx.picopbx.nl;tag=escuiu6jhn
To: sip:101@pbx.picopbx.nl;user=phone
Call-ID: 47044A553432012313A8628FBE846630-7nev3hg2uc1t
CSeq: 2 INVITE
Max-Forwards: 70
User-Agent: snom715/8.7.5.8.2
Contact: sip:pico-02@[2001:980:7936:0:204:13ff:fe75:5744]:32789;transport=tls;line=mjvft1a2;reg-id=1
X-Serialnumber: 000413755744
P-Key-Flags: resolution=“31x13”, keys=“4"
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Authorization: Digest username=“pico-02”,realm=“asterisk”,nonce=“12bbae7b”,uri="sip:101@pbx.picopbx.nl;user=phone”,response=“759fe7c63f5ef939dfe1fe357fdbb044”,algorithm=MD5
Content-Type: application/sdp
Content-Length: 729

v=0
o=root 1771457770 1771457770 IN IP6 2001:980:7936:0:204:13ff:fe75:5744
s=call
c=IN IP6 2001:980:7936:0:204:13ff:fe75:5744
t=0 0
m=audio 61136 RTP/SAVP 9 0 8 3 97 98 99 100 112 113 114 115 18 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:uOx0mT6szY6Iy2/yfz+Myl12OjzBFbcxLqOL/AS8
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 G726-16/8000
a=rtpmap:98 G726-24/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:100 G726-40/8000
a=rtpmap:112 AAL2-G726-16/8000
a=rtpmap:113 AAL2-G726-24/8000
a=rtpmap:114 AAL2-G726-32/8000
a=rtpmap:115 AAL2-G726-40/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
— (20 headers 25 lines) —
Sending to [2001:980:7936:0:204:13ff:fe75:5744]:32789 (no NAT)
Using INVITE request as basis request - 47044A553432012313A8628FBE846630-7nev3hg2uc1t
Found peer ‘pico-02’ for ‘pico-02’ from [2001:980:7936:0:204:13ff:fe75:5744]:32789
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 99
Found RTP audio format 100
Found RTP audio format 112
Found RTP audio format 113
Found RTP audio format 114
Found RTP audio format 115
Found RTP audio format 18
Found RTP audio format 101
[May 6 14:08:41] WARNING[8425][C-00000521]: sdp_srtp.c:187 crypto_activate: Could not set SRTP policies
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found unknown media description format G726-16 for ID 97
Found unknown media description format G726-24 for ID 98
Found audio description format G726-32 for ID 99
Found unknown media description format G726-40 for ID 100
Found unknown media description format AAL2-G726-16 for ID 112
Found unknown media description format AAL2-G726-24 for ID 113
Found audio description format AAL2-G726-32 for ID 114
Found unknown media description format AAL2-G726-40 for ID 115
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
[May 6 14:08:41] WARNING[8425][C-00000521]: chan_sip.c:10463 process_sdp: Rejecting secure audio stream without encryption details: audio 61136 RTP/SAVP 9 0 8 3 97 98 99 100 112 113 114 115 18 101

<— Reliably Transmitting (no NAT) to [2001:980:7936:0:204:13ff:fe75:5744]:32789 —>
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/TLS [2001:980:7936:0:204:13ff:fe75:5744]:32789;branch=z9hG4bK-mw6cgfvuxl6i;received=2001:980:7936:0:204:13ff:fe75:5744;rport=32789
From: “pico two” sip:pico-02@pbx.picopbx.nl;tag=escuiu6jhn
To: sip:101@pbx.picopbx.nl;user=phone;tag=as539b87ab
Call-ID: 47044A553432012313A8628FBE846630-7nev3hg2uc1t
CSeq: 2 INVITE
Server: Asterisk PBX 13.3.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘47044A553432012313A8628FBE846630-7nev3hg2uc1t’ in 6400 ms (Method: INVITE)

-------WARNING[8425][C-00000521]: chan_sip.c:10463 process_sdp: Rejecting secure audio stream without encryption details: audio 61136 RTP/SAVP 9 0 8 3 97 98 99 100 112 113 114 115 18 101
-------[May 6 14:08:41] WARNING[8425][C-00000521]: sdp_srtp.c:187 crypto_activate: Could not set SRTP policies

What do you make of this error?

Thanks