[asterisk-bugs] [JIRA] (ASTERISK-24291) res_srtp module stops working after about 35.000 processed calls

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24291?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=222614#comment-222614 ] 

Rusty Newton edited comment on ASTERISK-24291 at 9/11/14 1:20 PM:
------------------------------------------------------------------

Your trace didn't have RTP debug enabled, however doing so with your call volume could be massive (depending on your concurrent call state).

If you can get that and a PCAP showing those last few calls including the first failure, that would be excellent.  This is a strange issue.

Oh, additionally

 * configuration for the extensions involved. That is, sip.conf.

P.S. If your scenario is really a load test and not a production system, could you provide us all the configuration you are using with instructions to perform your load test so that we could reproduce it?


was (Author: rnewton):
Your trace didn't have RTP debug enabled, however doing so with your call volume could be massive (depending on your concurrent call state).

If you can get that and a PCAP showing those last few calls including the first failure, that would be excellent.  This is a strange issue.

Oh, additionally

 * configuration for the extensions involved. That is, sip.conf.

> res_srtp module stops working after about 35.000 processed calls
> ----------------------------------------------------------------
>
>                 Key: ASTERISK-24291
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24291
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/SRTP, Resources/res_srtp
>    Affects Versions: 11.11.0, 11.12.0
>         Environment: Ubuntu 12.04.5 LTS (GNU/Linux 3.13.0-34-generic x86_64) running on HP DL360 G6/7, latest  libsrtp0 version 1.4.4+20100615~dfsg-1build, SIP only environment
>            Reporter: Robert H.
>            Assignee: Matt Jordan
>            Severity: Critical
>         Attachments: issue_24291_full_log.14.txt
>
>
> When using encryption for RTP streams, asterisk does not accept any calls after about 35k calls (reproducable) have been processed correctly.
> All further inbound and outbound calls are rejected with a 
> "488 - Not Acceptable Here".
> When this happens, one asterisk machine shows:
> {noformat}
> [2014-08-29 17:32:23.807] DEBUG[28500][C-00009387]: chan_sip.c:10530 process_sdp: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:cYswzW2zYpdgsVkKgQWvdbUSLedzlE8nByMqEYiI... UNSUPPORTED OR FAILED.
> [2014-08-29 17:32:23.807] WARNING[28500][C-00009387]: chan_sip.c:10535 process_sdp: Rejecting secure audio stream without encryption details: audio 11070 RTP/SAVP 8 0 101
> {noformat}
> the destination asterisk shows:
> {noformat}
> WARNING[10222][C-0000883a]: chan_sip.c:12925 get_crypto_attrib: No SRTP key management enabled
> {noformat}
> Active srtp calls are not affected when this issue occurs, but all further Invites are rejected with the 488 response, so no more calls can be processed.
> The only solution at the moment is to restart asterisk or to wait until no more SRTP calls are active and then unload res_srtp.so followed by loading the module again.
> add info:
> - problem occurs regardless of using SIP over TLS or SIP without TLS
> - other (unencrypted) RTP connections are still working  
> If you need further info, just let me know.
> Thanks for checking into this!
> Robert
>     



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