[asterisk-bugs] [JIRA] (ASTERISK-28564) Memory leak with pjsip 2.9 and SIPS / SRTP

Michael Maier (JIRA) noreply at issues.asterisk.org
Sun Oct 20 13:25:49 CDT 2019


     [ https://issues.asterisk.org/jira/browse/ASTERISK-28564?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Michael Maier updated ASTERISK-28564:
-------------------------------------

    Send back to Developer?: I'm done. Send it back!  (was: I'm not done! I will comment again later to send it back.)
                     Status: Waiting for Feedback  (was: Waiting for Feedback)

The problem is tricky - most of the time, it works as it should, but in two cases e.g. I could see a mysterious rise of 1.1 MB during one hour (there was a call ongoing during those hours - but that can't be seen always). At the moment, the memory usage is 89 MByte. That's ok for my use case. Anyway, the reason for the sudden rises of the memory usage would be interesting.
Another reason for repeatedly (but not always) rising memory usage is sending or receiving of faxes (spandsp).
Maybe that's expected behavior? I can't say. If you want, you may close this ticket.

> Memory leak with pjsip 2.9 and SIPS / SRTP
> ------------------------------------------
>
>                 Key: ASTERISK-28564
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28564
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip
>    Affects Versions: 16.5.0
>         Environment: CentOS 7
>            Reporter: Michael Maier
>            Assignee: Michael Maier
>              Labels: pjsip
>
> There is one more memory leak even in asterisk 16.6.0-rc2, which can't be seen with pjsip 2.8 instead of 2.9. 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:+491234567890 at 12.13.14.15:5061;transport=tcp;line=abcdefg SIP/2.0
> Max-Forwards: 49
> Via: SIP/2.0/TLS 217.0.20.195:5061;branch=z9hG4bKg3Zqkv7ivdsp3wo1jhdbdvgy5dwsq6jye
> To: <sip:+491234567890 at telekom.de;user=phone>
> From: <sip:+4945678901234 at tmobile.de;user=phone>;tag=h7g4Esbg_p65540t1570108521m378032c299263169s1_1621954413-1461120854
> Call-ID: p65540t1570108521m378032c299263169s2
> CSeq: 1 INVITE
> Contact: <sip:sgc_c at 217.0.20.195:5061;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;rn=+49199C901234567890 at tmobile.de;user=phone>;index=1
> Min-Se: 900
> P-Asserted-Identity: <sip:+4945678901234 at tmobile.de;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:+4945678901234 at tmobile.de;user=phone>;tag=h7g4Esbg_p65540t1570108521m378032c299263169s1_1621954413-1461120854
> To: <sip:+491234567890 at telekom.de;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



--
This message was sent by Atlassian JIRA
(v6.2#6252)



More information about the asterisk-bugs mailing list