SIP debug log of a TLS/SRTP-call from subscriber 2004 to 2000 using a very recent Asterisk and PJSIP-stack client. Of note: - I sanitized domain and authentication data, so this can be transmitted publicly - The troublesome reply by Asterisk can be found in line 189 - The PJSIP-stack-based client BYE and error message is found in line 213 right behind it - The call is terminated as soon as 2000 picks up - PJSIP stacks before changeset 4899 ignore line 189 being sip: instead of sips: Connected to Asterisk SVN-branch-11-r430126M currently running on gateway (pid = 15596) gateway*CLI> sip set debug peer 2004 SIP Debugging Enabled for IP: 192.168.129.114 <--- SIP read from TLS:192.168.129.114:47371 ---> INVITE sips:2000@sip.deleted.de SIP/2.0 Via: SIP/2.0/TLS 192.168.129.114:47371;rport;branch=z9hG4bKPjKeuiES6R82bMGXkywlqBqnepNvg3LCiL;alias Max-Forwards: 70 From: ;tag=KKhbibQqmTegCdLU-no0CRSKC2y3gbaR To: Contact: Call-ID: QdH4NWTKbppR3rkL.m0uTLZ60AgpjxE- CSeq: 13777 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: CSipSimple_leo-17/r2456 Content-Type: application/sdp Content-Length: 520 v=0 o=- 3629874714 3629874714 IN IP4 192.168.129.114 s=pjmedia c=IN IP4 192.168.129.114 t=0 0 m=audio 4004 RTP/SAVP 9 18 105 101 c=IN IP4 192.168.129.114 a=rtcp:4005 IN IP4 192.168.129.114 a=sendrecv a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:105 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:deleted a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:deleted <-------------> --- (15 headers 17 lines) --- Sending to 192.168.129.114:47371 (NAT) Sending to 192.168.129.114:47371 (NAT) Using INVITE request as basis request - QdH4NWTKbppR3rkL.m0uTLZ60AgpjxE- Found peer '2004' for '2004' from 192.168.129.114:47371 <--- Reliably Transmitting (NAT) to 192.168.129.114:47371 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 192.168.129.114:47371;branch=z9hG4bKPjKeuiES6R82bMGXkywlqBqnepNvg3LCiL;alias;received=192.168.129.114;rport=47371 From: ;tag=KKhbibQqmTegCdLU-no0CRSKC2y3gbaR To: ;tag=as6d01a1d9 Call-ID: QdH4NWTKbppR3rkL.m0uTLZ60AgpjxE- CSeq: 13777 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6fddc7a5" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'QdH4NWTKbppR3rkL.m0uTLZ60AgpjxE-' in 19776 ms (Method: INVITE) <--- SIP read from TLS:192.168.129.114:47371 ---> ACK sips:2000@sip.deleted.de SIP/2.0 Via: SIP/2.0/TLS 192.168.129.114:47371;rport;branch=z9hG4bKPjKeuiES6R82bMGXkywlqBqnepNvg3LCiL;alias Max-Forwards: 70 From: ;tag=KKhbibQqmTegCdLU-no0CRSKC2y3gbaR To: ;tag=as6d01a1d9 Call-ID: QdH4NWTKbppR3rkL.m0uTLZ60AgpjxE- CSeq: 13777 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from TLS:192.168.129.114:47371 ---> INVITE sips:2000@sip.deleted.de SIP/2.0 Via: SIP/2.0/TLS 192.168.129.114:47371;rport;branch=z9hG4bKPj0tvwR82jNrNSqvCQ0ww9QV.IZQGCtjr7;alias Max-Forwards: 70 From: ;tag=KKhbibQqmTegCdLU-no0CRSKC2y3gbaR To: Contact: Call-ID: QdH4NWTKbppR3rkL.m0uTLZ60AgpjxE- CSeq: 13778 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: CSipSimple_leo-17/r2456 Authorization: Digest username="2004", realm="asterisk", nonce="6fddc7a5", uri="sips:2000@sip.deleted.de", response="deleted", algorithm=MD5 Content-Type: application/sdp Content-Length: 520 v=0 o=- 3629874714 3629874714 IN IP4 192.168.129.114 s=pjmedia c=IN IP4 192.168.129.114 t=0 0 m=audio 4004 RTP/SAVP 9 18 105 101 c=IN IP4 192.168.129.114 a=rtcp:4005 IN IP4 192.168.129.114 a=sendrecv a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:105 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:deleted a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:deleted <-------------> --- (16 headers 17 lines) --- Sending to 192.168.129.114:47371 (NAT) Using INVITE request as basis request - QdH4NWTKbppR3rkL.m0uTLZ60AgpjxE- Found peer '2004' for '2004' from 192.168.129.114:47371 Found RTP audio format 9 Found RTP audio format 18 Found RTP audio format 105 Found RTP audio format 101 Found audio description format G722 for ID 9 Found audio description format G729 for ID 18 Found audio description format speex for ID 105 Found audio description format telephone-event for ID 101 Capabilities: us - (gsm|alaw|g729|speex|speex16|g722|speex32), peer - audio=(g729|speex|g722)/video=(nothing)/text=(nothing), combined - (g729|speex|g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.129.114:4004 Looking for 2000 in outgoing (domain sip.deleted.de) list_route: hop: <--- Transmitting (NAT) to 192.168.129.114:47371 ---> SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.129.114:47371;branch=z9hG4bKPj0tvwR82jNrNSqvCQ0ww9QV.IZQGCtjr7;alias;received=192.168.129.114;rport=47371 From: ;tag=KKhbibQqmTegCdLU-no0CRSKC2y3gbaR To: Call-ID: QdH4NWTKbppR3rkL.m0uTLZ60AgpjxE- CSeq: 13778 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> <--- Transmitting (NAT) to 192.168.129.114:47371 ---> SIP/2.0 180 Ringing Via: SIP/2.0/TLS 192.168.129.114:47371;branch=z9hG4bKPj0tvwR82jNrNSqvCQ0ww9QV.IZQGCtjr7;alias;received=192.168.129.114;rport=47371 From: ;tag=KKhbibQqmTegCdLU-no0CRSKC2y3gbaR To: ;tag=as1891ca92 Call-ID: QdH4NWTKbppR3rkL.m0uTLZ60AgpjxE- CSeq: 13778 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> Audio is at 51572 Adding codec 100008 (g729) to SDP Adding codec 100009 (speex) to SDP Adding codec 100012 (g722) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 192.168.129.114:47371 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 192.168.129.114:47371;branch=z9hG4bKPj0tvwR82jNrNSqvCQ0ww9QV.IZQGCtjr7;alias;received=192.168.129.114;rport=47371 From: ;tag=KKhbibQqmTegCdLU-no0CRSKC2y3gbaR To: ;tag=as1891ca92 Call-ID: QdH4NWTKbppR3rkL.m0uTLZ60AgpjxE- CSeq: 13778 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 416 v=0 o=Administrator 1783657566 1783657566 IN IP4 192.168.129.1 s=Asterisk c=IN IP4 192.168.129.1 t=0 0 m=audio 51572 RTP/SAVP 18 105 9 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:105 speex/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:deleted <------------> <--- SIP read from TLS:192.168.129.114:47371 ---> BYE sip:2000@192.168.129.1:5061;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 192.168.129.114:47371;rport;branch=z9hG4bKPjasBZgNK3nGZ0iXlyrGcRgW-Ew.jjkXuY;alias Max-Forwards: 70 From: ;tag=KKhbibQqmTegCdLU-no0CRSKC2y3gbaR To: ;tag=as1891ca92 Call-ID: QdH4NWTKbppR3rkL.m0uTLZ60AgpjxE- CSeq: 13779 BYE Warning: 381 localhost "SIPS Required" Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- Reliably Transmitting (NAT) to 192.168.129.114:47371 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/TLS 192.168.129.114:47371;branch=z9hG4bKPj0tvwR82jNrNSqvCQ0ww9QV.IZQGCtjr7;alias;received=192.168.129.114;rport=47371 From: ;tag=KKhbibQqmTegCdLU-no0CRSKC2y3gbaR To: ;tag=as1891ca92 Call-ID: QdH4NWTKbppR3rkL.m0uTLZ60AgpjxE- CSeq: 13778 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> Sending to 192.168.129.114:47371 (NAT) Scheduling destruction of SIP dialog 'QdH4NWTKbppR3rkL.m0uTLZ60AgpjxE-' in 19776 ms (Method: BYE) <--- Transmitting (NAT) to 192.168.129.114:47371 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 192.168.129.114:47371;branch=z9hG4bKPjasBZgNK3nGZ0iXlyrGcRgW-Ew.jjkXuY;alias;received=192.168.129.114;rport=47371 From: ;tag=KKhbibQqmTegCdLU-no0CRSKC2y3gbaR To: ;tag=as1891ca92 Call-ID: QdH4NWTKbppR3rkL.m0uTLZ60AgpjxE- CSeq: 13779 BYE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------>