[asterisk-bugs] [JIRA] Commented: (ASTERISK-20194) SRTP: after hold action no audio on holded peer.

[Matt Jordan (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-20194?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=195576#comment-195576 ] 

Matt Jordan commented on ASTERISK-20194:
----------------------------------------

The logs here are rather confusing.

In the SIP pcap trace, the UA in question sends the same crypto key in the INVITE requests.  There is nothing wrong with this - doing so merely causes Asterisk to reload the same policy that it previously had esablished for that SSRC.  Immediately after taking the call off of hold, however, the RTP packet unauthenticates fail.  This would imply one of two things:

1) The UA is not honoring the key it sent us
2) The UA has changed the SSRC in the RTP stream

What I can't see from the pcaps is whether or not the second condition is the case.  Can you include a pcap that also includes the RTCP/SRTP packets?  That will let me know whether or not the second case is the cause of this.

> SRTP: after hold action no audio on holded peer.
> ------------------------------------------------
>
>                 Key: ASTERISK-20194
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20194
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_srtp
>    Affects Versions: 1.8.13.1, 1.8.14.0, 1.8.15.0
>         Environment: Linux Centos 5.7
>            Reporter: Nicolò Mazzon
>            Assignee: Matt Jordan
>            Severity: Critical
>         Attachments: config.txt, full.log, full.tmp, sip_trace.pcap, siptrace.pcap
>
>
> On a call with SRTP after peer hold another and unhold, the holded peer does not hear anything, the holder peer continue to hear audio call. This happens after 10-15 minutes of conversation. 
> We verified this every time with version from version 1.8.13. Instead 1.8.12 works ok.
> We made many tests with different phone models (SNOM, Polycom, Yealink) and issue occured every time.
> We attach our configuration and log. Log level is verbose = 12 and debug = 12.
> This is the scenario:
> ||action||hour||
> |2209 call 2210| 10.43|
> |2210 answer||                     
> |2210 hold|11.00|
> |2210 unhold|11.01|
> |2209 no audio||
> |2209 hangup up|11.02|

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