[asterisk-bugs] [JIRA] (ASTERISK-28564) Memory leak with pjsip 2.9 and SIPS / SRTP
Michael Maier (JIRA)
noreply at issues.asterisk.org
Mon Oct 7 14:45:47 CDT 2019
[ https://issues.asterisk.org/jira/browse/ASTERISK-28564?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=248285#comment-248285 ]
Michael Maier edited comment on ASTERISK-28564 at 10/7/19 2:44 PM:
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I measured it on base of of ps aux / RSS. I can't see any growth on RSS using pjsip 2.8, but 2.9 (same asterisk version 16.5.1 e.g.).
Maybe packet sizes are different? The udp packet size here is 226 bytes (length as shown by wireshark) and data length is 182 bytes.
Unfortunately, I'm not able to do any further investigation at the moment. Maybe next weekend (but not sure).
Please don't spend unnecessary time on it as long as you don't have any clever idea by chance.
Is there something like git bisect for the pjsip code? Maybe I can find the culprit this way? Would be more easily I think.
was (Author: micha):
I measured it on base of base of ps aux / RSS. I can't see any growth on RSS using pjsip 2.8, but 2.9 (same asterisk version 16.5.1 e.g.).
Maybe packet sizes are different? The udp packet size here is 226 bytes (length as shown by wireshark) and data length is 182 bytes.
Unfortunately, I'm not able to do any further investigation at the moment. Maybe next weekend (but not sure).
Please don't spend unnecessary time on it as long as you don't have any clever idea by chance.
Is there something like git bisect for the pjsip code? Maybe I can find the culprit this way? Would be more easily I think.
> 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: Joshua C. Colp
> 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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