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

Thu Oct 3 09:19:47 CDT 2019

    [ https://issues.asterisk.org/jira/browse/ASTERISK-28564?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=248222#comment-248222 ] 

Joshua C. Colp edited comment on ASTERISK-28564 at 10/3/19 9:18 AM:
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[~micha] [~seanbright] is aware of SRTP / TLS configuration but we've found that in general having a specific configuration reduces the amount of time wasted on trying to track things down due to differences, so it's standard practice to ask for such things. I once spent over a week trying to find a problem waiting for a reporter to respond, once they did and provided their configuration I was able to track it down in minutes. That's why we treat it as standard practice. You are certainly free not to provide it and we still attempt to isolate and identify problems.

was (Author: jcolp):
[~micha] [~seanbright] is aware of SRTP / TLS configuration but we've found that in general having a specific configuration reduces the amount of time wasted on trying to track things down due to differences, so it's standard practice to ask for such things. I once spent over a week trying to find a problem waiting for a reporter to respond, once they did and provided their configuration I was able to track it down in minutes. That's why we treat it as standard practice.

> 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: Unassigned
>              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

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