Failed to receive SDP offer

Hello,

I am trying to install an asterisk pbx server with TLS, SRTP, Directmedia all on IPv6.
SIP TLS is working prefect, so is direct media. The problem is when i try to activate SRTP.
I have searched the internet for some a solution with no success.

sip log

[code]<— SIP read from TLS:[2001:980:7936:0:204:13ff:fe75:5744]:48245 —>
INVITE sip:101@pbx.picopbx.nl;user=phone SIP/2.0
Via: SIP/2.0/TLS [2001:980:7936:0:204:13ff:fe75:5744]:48245;branch=z9hG4bK-gptn16ofr4oz;rport
From: “two” sip:pico-02@pbx.picopbx.nl;tag=s4kzldvldj
To: sip:101@pbx.picopbx.nl;user=phone
Call-ID: 3C654B55D9F1091D9FA8B2A2BE846630-ajnj2bhh02ix
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]:48245;transport=tls;line=u0fomw11;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: 1270

v=0
o=root 315126636 315126636 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 61810 RTP/SAVP 9 0 8 3 97 98 99 100 112 113 114 115 18 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:GEqDeZgItBxADhyS85+8tAYpfKVgl8TqUtUfCyNK
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
m=audio 61810 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 46 lines) —
Sending to [2001:980:7936:0:204:13ff:fe75:5744]:48245 (no NAT)
Sending to [2001:980:7936:0:204:13ff:fe75:5744]:48245 (no NAT)
Using INVITE request as basis request - 3C654B55D9F1091D9FA8B2A2BE846630-ajnj2bhh02ix
Found peer ‘pico-02’ for ‘pico-02’ from [2001:980:7936:0:204:13ff:fe75:5744]:48245

<— Reliably Transmitting (no NAT) to [2001:980:7936:0:204:13ff:fe75:5744]:48245 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS [2001:980:7936:0:204:13ff:fe75:5744]:48245;branch=z9hG4bK-gptn16ofr4oz;received=2001:980:7936:0:204:13ff:fe75:5744;rport=48245
From: “two” sip:pico-02@pbx.picopbx.nl;tag=s4kzldvldj
To: sip:101@pbx.picopbx.nl;user=phone;tag=as2e166f39
Call-ID: 3C654B55D9F1091D9FA8B2A2BE846630-ajnj2bhh02ix
CSeq: 1 INVITE
Server: Asterisk PBX 11.17.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="1d0f27ab"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘3C654B55D9F1091D9FA8B2A2BE846630-ajnj2bhh02ix’ in 6400 ms (Method: INVITE)

<— SIP read from TLS:[2001:980:7936:0:204:13ff:fe75:5744]:48245 —>
ACK sip:101@pbx.picopbx.nl;user=phone SIP/2.0
Via: SIP/2.0/TLS [2001:980:7936:0:204:13ff:fe75:5744]:48245;branch=z9hG4bK-gptn16ofr4oz;rport
From: “two” sip:pico-02@pbx.picopbx.nl;tag=s4kzldvldj
To: sip:101@pbx.picopbx.nl;user=phone;tag=as2e166f39
Call-ID: 3C654B55D9F1091D9FA8B2A2BE846630-ajnj2bhh02ix
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]:48245;transport=tls;line=u0fomw11;reg-id=1
Content-Length: 0

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

<— SIP read from TLS:[2001:980:7936:0:204:13ff:fe75:5744]:48245 —>
INVITE sip:101@pbx.picopbx.nl;user=phone SIP/2.0
Via: SIP/2.0/TLS [2001:980:7936:0:204:13ff:fe75:5744]:48245;branch=z9hG4bK-a99185v850f2;rport
From: “two” sip:pico-02@pbx.picopbx.nl;tag=s4kzldvldj
To: sip:101@pbx.picopbx.nl;user=phone
Call-ID: 3C654B55D9F1091D9FA8B2A2BE846630-ajnj2bhh02ix
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]:48245;transport=tls;line=u0fomw11;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=“1d0f27ab”,uri="sip:101@pbx.picopbx.nl;user=phone”,response=“ea5e81f74522af250def829e4c45dcc2”,algorithm=MD5
Content-Type: application/sdp
Content-Length: 1270

v=0
o=root 315126636 315126636 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 61810 RTP/SAVP 9 0 8 3 97 98 99 100 112 113 114 115 18 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:GEqDeZgItBxADhyS85+8tAYpfKVgl8TqUtUfCyNK
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
m=audio 61810 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 46 lines) —
Sending to [2001:980:7936:0:204:13ff:fe75:5744]:48245 (no NAT)
Using INVITE request as basis request - 3C654B55D9F1091D9FA8B2A2BE846630-ajnj2bhh02ix
Found peer ‘pico-02’ for ‘pico-02’ from [2001:980:7936:0:204:13ff:fe75:5744]:48245
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 7 15:53:29] WARNING[24767][C-00000015]: chan_sip.c:9984 process_sdp: Declining non-primary audio stream: audio 61810 RTP/AVP 9 0 8 3 97 98 99 100 112 113 114 115 18 101
Capabilities: us - (gsm|ulaw|alaw|g722|h263|testlaw), peer - audio=(gsm|ulaw|alaw|g726|g729|g726aal2|g722)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port [2001:980:7936:0:204:13ff:fe75:5744]:61810
Peer doesn’t provide video
Looking for 101 in intern (domain pbx.picopbx.nl)
list_route: hop: sip:pico-02@[2001:980:7936:0:204:13ff:fe75:5744]:48245;transport=tls;line=u0fomw11

<— Transmitting (no NAT) to [2001:980:7936:0:204:13ff:fe75:5744]:48245 —>
SIP/2.0 100 Trying
Via: SIP/2.0/TLS [2001:980:7936:0:204:13ff:fe75:5744]:48245;branch=z9hG4bK-a99185v850f2;received=2001:980:7936:0:204:13ff:fe75:5744;rport=48245
From: “two” sip:pico-02@pbx.picopbx.nl;tag=s4kzldvldj
To: sip:101@pbx.picopbx.nl;user=phone
Call-ID: 3C654B55D9F1091D9FA8B2A2BE846630-ajnj2bhh02ix
CSeq: 2 INVITE
Server: Asterisk PBX 11.17.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:101@[2a00:f10:121:400:4ce:48ff:fe00:1410]:5061;transport=TLS
Content-Length: 0

<------------>
Audio is at 12068
Video is at [2a00:f10:121:400:4ce:48ff:fe00:1410]:13474
Adding codec 100012 (g722) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100017 (testlaw) to SDP
Adding video codec 200002 (h263) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to [2001:980:7936:0:204:13ff:fe75:5737]:52359:
INVITE sip:pico-01@[2001:980:7936:0:204:13ff:fe75:5737]:52359;transport=tls;line=912j203t SIP/2.0
Via: SIP/2.0/TLS [2a00:f10:121:400:4ce:48ff:fe00:1410]:5061;branch=z9hG4bK4da54a94
Max-Forwards: 70
From: “two” sip:pico-02@[2a00:f10:121:400:4ce:48ff:fe00:1410];tag=as512d1db5
To: sip:pico-01@[2001:980:7936:0:204:13ff:fe75:5737]:52359;transport=tls;line=912j203t
Contact: sip:pico-02@[2a00:f10:121:400:4ce:48ff:fe00:1410]:5061;transport=TLS
Call-ID: 0f260ed87d3073b30687bb1b68312b4a@[2a00:f10:121:400:4ce:48ff:fe00:1410]:5061
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.17.0
Date: Thu, 07 May 2015 13:53:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 796

v=0
o=root 249155712 249155712 IN IP6 2a00:f10:121:400:4ce:48ff:fe00:1410
s=Asterisk PBX 11.17.0
c=IN IP6 2a00:f10:121:400:4ce:48ff:fe00:1410
b=CT:384
t=0 0
m=audio 12068 UDP/TLS/RTP/SAVP 9 0 8 3 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 15:FB:EA:F4:87:D7:91:B6:C8:98:9C:D0:EE:A8:94:F3:93:84:BF:BD:30:B4:68:B8:0D:A7:B2:92:B7:A3:E9:37
a=sendrecv
m=video 13474 UDP/TLS/RTP/SAVP 34
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 15:FB:EA:F4:87:D7:91:B6:C8:98:9C:D0:EE:A8:94:F3:93:84:BF:BD:30:B4:68:B8:0D:A7:B2:92:B7:A3:E9:37
a=rtpmap:34 H263/90000
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=sendrecv


<— SIP read from TLS:[2001:980:7936:0:204:13ff:fe75:5737]:52359 —>
SIP/2.0 100 Trying
Via: SIP/2.0/TLS [2a00:f10:121:400:4ce:48ff:fe00:1410]:5061;branch=z9hG4bK4da54a94;received=2a00:f10:121:400:4ce:48ff:fe00:1410
From: “two” sip:pico-02@[2a00:f10:121:400:4ce:48ff:fe00:1410];tag=as512d1db5
To: sip:pico-01@[2001:980:7936:0:204:13ff:fe75:5737]:52359;transport=tls;line=912j203t;tag=yau6laxoif
Call-ID: 0f260ed87d3073b30687bb1b68312b4a@[2a00:f10:121:400:4ce:48ff:fe00:1410]:5061
CSeq: 102 INVITE
User-Agent: snom715/8.7.5.8.2
Contact: sip:pico-01@[2001:980:7936:0:204:13ff:fe75:5737]:52359;transport=tls;line=912j203t;reg-id=1
Content-Length: 0

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

<— SIP read from TLS:[2001:980:7936:0:204:13ff:fe75:5737]:52359 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS [2a00:f10:121:400:4ce:48ff:fe00:1410]:5061;branch=z9hG4bK4da54a94;received=2a00:f10:121:400:4ce:48ff:fe00:1410
From: “two” sip:pico-02@[2a00:f10:121:400:4ce:48ff:fe00:1410];tag=as512d1db5
To: sip:pico-01@[2001:980:7936:0:204:13ff:fe75:5737]:52359;transport=tls;line=912j203t;tag=yau6laxoif
Call-ID: 0f260ed87d3073b30687bb1b68312b4a@[2a00:f10:121:400:4ce:48ff:fe00:1410]:5061
CSeq: 102 INVITE
User-Agent: snom715/8.7.5.8.2
Contact: sip:pico-01@[2001:980:7936:0:204:13ff:fe75:5737]:52359;transport=tls;line=912j203t;reg-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Content-Length: 0

<------------->
— (11 headers 0 lines) —
list_route: hop: sip:pico-01@[2001:980:7936:0:204:13ff:fe75:5737]:52359;transport=tls;line=912j203t

<— Transmitting (no NAT) to [2001:980:7936:0:204:13ff:fe75:5744]:48245 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS [2001:980:7936:0:204:13ff:fe75:5744]:48245;branch=z9hG4bK-a99185v850f2;received=2001:980:7936:0:204:13ff:fe75:5744;rport=48245
From: “two” sip:pico-02@pbx.picopbx.nl;tag=s4kzldvldj
To: sip:101@pbx.picopbx.nl;user=phone;tag=as0d1f4ccc
Call-ID: 3C654B55D9F1091D9FA8B2A2BE846630-ajnj2bhh02ix
CSeq: 2 INVITE
Server: Asterisk PBX 11.17.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:101@[2a00:f10:121:400:4ce:48ff:fe00:1410]:5061;transport=TLS
Content-Length: 0

<------------>

<— SIP read from TLS:[2001:980:7936:0:204:13ff:fe75:5737]:52359 —>
SIP/2.0 200 Ok
Via: SIP/2.0/TLS [2a00:f10:121:400:4ce:48ff:fe00:1410]:5061;branch=z9hG4bK4da54a94;received=2a00:f10:121:400:4ce:48ff:fe00:1410
From: “two” sip:pico-02@[2a00:f10:121:400:4ce:48ff:fe00:1410];tag=as512d1db5
To: sip:pico-01@[2001:980:7936:0:204:13ff:fe75:5737]:52359;transport=tls;line=912j203t;tag=yau6laxoif
Call-ID: 0f260ed87d3073b30687bb1b68312b4a@[2a00:f10:121:400:4ce:48ff:fe00:1410]:5061
CSeq: 102 INVITE
User-Agent: snom715/8.7.5.8.2
Contact: sip:pico-01@[2001:980:7936:0:204:13ff:fe75:5737]:52359;transport=tls;line=912j203t;reg-id=1
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
Content-Type: application/sdp
Content-Length: 621

v=0
o=root 1283359601 1283359602 IN IP6 2001:980:7936:0:204:13ff:fe75:5737
s=call
c=IN IP6 2001:980:7936:0:204:13ff:fe75:5737
t=0 0
m=audio 63334 RTP/AVP 9 0 8 3 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=direction:both
a=sendrecv
m=video 0 UDP/TLS/RTP/SAVP 34
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 15:FB:EA:F4:87:D7:91:B6:C8:98:9C:D0:EE:A8:94:F3:93:84:BF:BD:30:B4:68:B8:0D:A7:B2:92:B7:A3:E9:37
a=rtpmap:34 H263/90000
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=sendrecv
<------------->
— (13 headers 22 lines) —
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
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 audio description format telephone-event for ID 101
[May 7 15:53:30] WARNING[24766][C-00000015]: chan_sip.c:10088 process_sdp: Ignoring video stream offer because port number is zero
[May 7 15:53:30] WARNING[24766][C-00000015]: chan_sip.c:10427 process_sdp: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio
list_route: hop: sip:pico-01@[2001:980:7936:0:204:13ff:fe75:5737]:52359;transport=tls;line=912j203t
set_destination: Parsing sip:pico-01@[2001:980:7936:0:204:13ff:fe75:5737]:52359;transport=tls;line=912j203t for address/port to send to
set_destination: set destination to [2001:980:7936:0:204:13ff:fe75:5737]:52359
Transmitting (no NAT) to [2001:980:7936:0:204:13ff:fe75:5737]:52359:
ACK sip:pico-01@[2001:980:7936:0:204:13ff:fe75:5737]:52359;transport=tls;line=912j203t SIP/2.0
Via: SIP/2.0/TLS [2a00:f10:121:400:4ce:48ff:fe00:1410]:5061;branch=z9hG4bK5670eafa
Max-Forwards: 70
From: “two” sip:pico-02@[2a00:f10:121:400:4ce:48ff:fe00:1410];tag=as512d1db5
To: sip:pico-01@[2001:980:7936:0:204:13ff:fe75:5737]:52359;transport=tls;line=912j203t;tag=yau6laxoif
Contact: sip:pico-02@[2a00:f10:121:400:4ce:48ff:fe00:1410]:5061;transport=TLS
Call-ID: 0f260ed87d3073b30687bb1b68312b4a@[2a00:f10:121:400:4ce:48ff:fe00:1410]:5061
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.17.0
Content-Length: 0


set_destination: Parsing sip:pico-01@[2001:980:7936:0:204:13ff:fe75:5737]:52359;transport=tls;line=912j203t for address/port to send to
set_destination: set destination to [2001:980:7936:0:204:13ff:fe75:5737]:52359
Reliably Transmitting (no NAT) to [2001:980:7936:0:204:13ff:fe75:5737]:52359:
BYE sip:pico-01@[2001:980:7936:0:204:13ff:fe75:5737]:52359;transport=tls;line=912j203t SIP/2.0
Via: SIP/2.0/TLS [2a00:f10:121:400:4ce:48ff:fe00:1410]:5061;branch=z9hG4bK7dbb665f
Max-Forwards: 70
From: “two” sip:pico-02@[2a00:f10:121:400:4ce:48ff:fe00:1410];tag=as512d1db5
To: sip:pico-01@[2001:980:7936:0:204:13ff:fe75:5737]:52359;transport=tls;line=912j203t;tag=yau6laxoif
Call-ID: 0f260ed87d3073b30687bb1b68312b4a@[2a00:f10:121:400:4ce:48ff:fe00:1410]:5061
CSeq: 103 BYE
User-Agent: Asterisk PBX 11.17.0
X-Asterisk-HangupCause: Bearer capability not available
X-Asterisk-HangupCauseCode: 58
Content-Length: 0


Scheduling destruction of SIP dialog ‘0f260ed87d3073b30687bb1b68312b4a@[2a00:f10:121:400:4ce:48ff:fe00:1410]:5061’ in 6400 ms (Method: INVITE)
Scheduling destruction of SIP dialog ‘0f260ed87d3073b30687bb1b68312b4a@[2a00:f10:121:400:4ce:48ff:fe00:1410]:5061’ in 6400 ms (Method: INVITE)

<— Reliably Transmitting (no NAT) to [2001:980:7936:0:204:13ff:fe75:5744]:48245 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/TLS [2001:980:7936:0:204:13ff:fe75:5744]:48245;branch=z9hG4bK-a99185v850f2;received=2001:980:7936:0:204:13ff:fe75:5744;rport=48245
From: “two” sip:pico-02@pbx.picopbx.nl;tag=s4kzldvldj
To: sip:101@pbx.picopbx.nl;user=phone;tag=as0d1f4ccc
Call-ID: 3C654B55D9F1091D9FA8B2A2BE846630-ajnj2bhh02ix
CSeq: 2 INVITE
Server: Asterisk PBX 11.17.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
X-Asterisk-HangupCause: Bearer capability not available
X-Asterisk-HangupCauseCode: 58
Content-Length: 0

<------------>

<— SIP read from TLS:[2001:980:7936:0:204:13ff:fe75:5744]:48245 —>
ACK sip:101@pbx.picopbx.nl;user=phone SIP/2.0
Via: SIP/2.0/TLS [2001:980:7936:0:204:13ff:fe75:5744]:48245;branch=z9hG4bK-a99185v850f2;rport
From: “two” sip:pico-02@pbx.picopbx.nl;tag=s4kzldvldj
To: sip:101@pbx.picopbx.nl;user=phone;tag=as0d1f4ccc
Call-ID: 3C654B55D9F1091D9FA8B2A2BE846630-ajnj2bhh02ix
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]:48245;transport=tls;line=u0fomw11;reg-id=1
Content-Length: 0

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

<— SIP read from TLS:[2001:980:7936:0:204:13ff:fe75:5737]:52359 —>
SIP/2.0 200 OK
Via: SIP/2.0/TLS [2a00:f10:121:400:4ce:48ff:fe00:1410]:5061;branch=z9hG4bK7dbb665f;received=2a00:f10:121:400:4ce:48ff:fe00:1410
From: “two” sip:pico-02@[2a00:f10:121:400:4ce:48ff:fe00:1410];tag=as512d1db5
To: sip:pico-01@[2001:980:7936:0:204:13ff:fe75:5737]:52359;transport=tls;line=912j203t;tag=yau6laxoif
Call-ID: 0f260ed87d3073b30687bb1b68312b4a@[2a00:f10:121:400:4ce:48ff:fe00:1410]:5061
CSeq: 103 BYE
User-Agent: snom715/8.7.5.8.2
Contact: sip:pico-01@[2001:980:7936:0:204:13ff:fe75:5737]:52359;transport=tls;line=912j203t;reg-id=1
RTP-RxStat: Total_Rx_Pkts=0,Rx-Pkts=0,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0
RTP-TxStat: Total_Tx_Pkts=0,Tx_Pkts=0,Remote_Tx_Pkts=0
Content-Length: 0
[/code]

snom client log

May  7 14:26:26  [NOTICE] PHN: TPL: Socket 198 idle/connect timeout
May  7 14:26:28  [WARN  ] SIP: process_registrar_packet: 401 needs 128 bit nonce
May  7 14:26:28  [NOTICE] SIP: process auth: Match challenge for user=pico-02, realm=asterisk
May  7 14:26:31  [NOTICE] MEDIA: MediaIpc::rtpClose: RP40
May  7 14:26:31  [NOTICE] MEDIA: MediaIpc::rtpClose: RC40
May  7 14:26:31  [ERROR ] MEDIA: Stream setup: invalid ssrc 1398009464, state off
May  7 14:26:31  [ERROR ] MEDIA: Stream close: invalid ssrc 1398009464
May  7 14:26:31  [NOTICE] MEDIA: onRtpClose: RP40
May  7 14:26:31  [NOTICE] MEDIA: onRtpClose: RP40 execute
May  7 14:26:31  [NOTICE] MEDIA: onRtpClose: RC40
May  7 14:26:31  [NOTICE] MEDIA: onRtpClose: RC40 execute

sip.conf

[general]
context=intern
allowoverlap=no
udpbindaddr=::
tcpenable=yes
tcpbindaddr=::
srvlookup=yes
qualify=yes
nat=no
transport=tls
host=dynamic
directmedia=yes
allow=g722
allow=ulaw
allow=alaw
encryption=yes

dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/pbx.picopbx.nl.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass

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

[pico-01](general)
type=peer
context=intern
secret=
srtpcapable=yes
[pico-02](general)
type=peer
context=intern
secret=
srtpcapable=yes

sip show peer

* Name : pico-01 Description : Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : intern Record On feature : automon Record Off feature : automon Subscr.Cont. : <Not set> Language : Tonezone : <Not set> AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : Named Callgr : Nam. Pickupgr: MOH Suggest : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Max forwards : 0 Dynamic : Yes Callerid : "" <> MaxCallBR : 384 kbps Expire : 2104 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: Yes Text Support : No Ign SDP ver : No Trust RPID : No Send RPID : No TrustIDOutbnd: Legacy Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : [2001:980:7936:0:204:13ff:fe75:5737]:52359 Defaddr->IP : (null) Prim.Transp. : TLS Allowed.Trsp : TLS Def. Username: pico-01 SIP Options : (none) Codecs : (gsm|ulaw|alaw|g722|h263|testlaw) Codec Order : (g722:20,ulaw:20,alaw:20) Auto-Framing : No Status : OK (33 ms) Useragent : snom715/8.7.5.8.2 Reg. Contact : sip:pico-01@[2001:980:7936:0:204:13ff:fe75:5737]:52359;transport=tls;line=912j203t 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 : Yes

Any help will be appreciated!! Thx alot!

I don’t believe Asterisk supports direct media with SRTP.

Whilst Asterisk doesn’t really handle SSRCs properly, I’ve never seen anything object in the way shown.