Hello again! Sorry, but there is one more memory leak even in asterisk 16.6.0-rc2, which can't be seen with pjsip 4.8 instead of 4.9. It can be seen on inbound calls (not sure if it's on outbound calls, too) using SIPS and SRTP.
Examples: 1 Call, duration about 1 h: ~ +1,2 MB 5 short calls (< 1 minute): ~ +1 MB Example for the inbound INVITE and OK package: <--- Received SIP request (2276 bytes) from TLS:217.0.20.195:5061 ---> INVITE sip:[email protected]:5061;transport=tcp;line=abcdefg SIP/2.0 Max-Forwards: 49 Via: SIP/2.0/TLS 217.0.20.195:5061;branch=z9hG4bKg3Zqkv7ivdsp3wo1jhdbdvgy5dwsq6jye To: <sip:[email protected];user=phone> From: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65540t1570108521m378032c299263169s1_1621954413-1461120854 Call-ID: p65540t1570108521m378032c299263169s2 CSeq: 1 INVITE Contact: <sip:[email protected]:5061;transport=tls>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" Record-Route: <sip:217.0.20.195:5061;transport=tls;lr> Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" History-Info: <sip:+491234567890;npdi;[email protected];user=phone>;index=1 Min-Se: 900 P-Asserted-Identity: <sip:[email protected];user=phone> P-Asserted-Identity: <tel:+4945678901234> Session-Expires: 1800 Supported: timer Supported: 100rel Supported: histinfo Supported: 199 Supported: uui Supported: norefersub Content-Type: application/sdp Content-Length: 1061 Session-ID: 253f41678c65f936805ef6b071943e64 Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, UPDATE, PRACK, INFO, INVITE, ACK, OPTIONS, CANCEL, BYE v=0 o=- 1011696818 1621954173 IN IP4 217.0.20.195 s=- c=IN IP4 217.0.135.5 t=0 0 m=audio 27888 RTP/SAVP 96 97 9 98 99 100 101 8 102 103 b=AS:84 a=rtpmap:96 AMR-WB/16000 a=fmtp:96 mode-set=0,1,2; mode-change-period=2; mode-change-neighbor=1; max-red=0 a=rtpmap:97 AMR-WB/16000 a=fmtp:97 mode-change-capability=2; max-red=0 a=rtpmap:9 G722/8000 a=rtpmap:98 AMR/8000 a=fmtp:98 mode-set=0,2,4,7; mode-change-period=2; mode-change-neighbor=1; max-red=0 a=rtpmap:99 AMR/8000 a=fmtp:99 mode-set=0,2,4; mode-change-period=2; mode-change-neighbor=1; max-red=0 a=rtpmap:100 AMR/8000 a=fmtp:100 mode-set=0,1,2,3,4,5,6,7; mode-change-period=2; mode-change-neighbor=1; max-red=0 a=rtpmap:101 AMR/8000 a=fmtp:101 mode-set=0,1,2,3,4,5,6,7; max-red=0 a=rtpmap:8 PCMA/8000 a=rtpmap:102 telephone-event/8000 a=rtpmap:103 telephone-event/16000 a=ptime:20 a=maxptime:30 a=3ge2ae:applied a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:HTNhK8lOYS+/1ORuNEbEhnsisXj4PEVIh8FBKmTR a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:6qBJEfKKXxbpJTepS298yUmUl/891GwnlURC3tdn <--- Transmitting SIP response (1178 bytes) to TLS:217.0.20.195:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 217.0.20.195:5061;rport=5061;received=217.0.20.195;branch=z9hG4bKg3Zqkv7ivdsp3wo1jhdbdvgy5dwsq6jye Record-Route: <sip:217.0.20.195:5061;transport=TLS;lr> Call-ID: p65540t1570108521m378032c299263169s2 From: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65540t1570108521m378032c299263169s1_1621954413-1461120854 To: <sip:[email protected];user=phone>;tag=94f22858-9c32-44c5-8a45-76964f62684a CSeq: 1 INVITE Server: FPBX-14.0.11(16.5.1) Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Contact: <sip:12.13.14.15:5061;transport=TLS> Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800;refresher=uac Require: timer Content-Type: application/sdp Content-Length: 368 v=0 o=- 1011696818 1621954176 IN IP4 12.13.14.15 s=Asterisk c=IN IP4 12.13.14.15 t=0 0 m=audio 10032 RTP/SAVP 9 8 102 a=3ge2ae:requested a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:OnkHAdHasSl83UnyFNuDSrBx+OsRF8DRZ6c5PnmJ a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:102 telephone-event/8000 a=fmtp:102 0-16 a=ptime:20 a=maxptime:150 a=sendrecv Thanks Michael -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
