[asterisk-bugs] [JIRA] (ASTERISK-24657) directmedia=no does not work in Asterisk 13.1.0

[Thomas B. Clark (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24657?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=224280#comment-224280 ] 

Thomas B. Clark commented on ASTERISK-24657:
--------------------------------------------

Thanks for the explanation of the native_rtp module; I was not aware of that.  However, I think it did issue reinvites.  I did not them in the debug messages, but the native_rtp causes one-way audio, which it would have no reason to do if the pathway were still through Asterisk.  Also adding ,,t to the Dial command fixes the one-way audio, which it likely does by preventing the external reinvites.  I would be glad to do more testing if you will tell me what you would like.

> directmedia=no does not work in Asterisk 13.1.0
> -----------------------------------------------
>
>                 Key: ASTERISK-24657
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24657
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 13.1.0
>         Environment: The Asterisk server runs under Fedora 20, with static ipv4 addresses and working ipv6.
> The yealinkphone is a sip client of the asterisk server.  The yealink phone has a non-routable ipv4 address and a routable ipv6 address, but preferentially uses the ipv6 address.
> The service provider, callwithus, does not have an ipv6 address.
>            Reporter: Thomas B. Clark
>            Assignee: Thomas B. Clark
>
> I just upgraded from Asterisk 11.14.2 to asterisk 13.1.0, and that caused directmedia=no to stop working.  It worked in 11.14, and using the same sip.conf it does not work in 13.1.0.  Here is the relevant part of sip.conf:
> [yealinkphone]
> type=friend
> host=dynamic
> context=outgoing
> secret=xxxxxx
> defaultuser=xxxxxxx
> insecure=port,invite
> directmedia=no
> qualify=yes
> And here is what happens:
>     -- SIP/callwithus-00000003 is making progress passing it to SIP/yealinkphone-00000002
>        > 0x7fa784016470 -- Probation passed - setting RTP source address to 69.85.185.222:21708
>     -- SIP/callwithus-00000003 answered SIP/yealinkphone-00000002
> -- Channel SIP/yealinkphone-00000002 joined 'simple_bridge' basic-bridge <5fb81915-9eaa-4b69-9be8-26f8a1db1fff>
>     -- Channel SIP/callwithus-00000003 joined 'simple_bridge' basic-bridge <5fb81915-9eaa-4b69-9be8-26f8a1db1fff>
>        > Bridge 5fb81915-9eaa-4b69-9be8-26f8a1db1fff: switching from simple_bridge technology to native_rtp
>        > 0x7fa70400afd0 -- Probation passed - setting RTP source address to [2601:8:9181:4800:215:65ff:fe27:ac8e]:11786
> Even if I had directmedia=yes, Asterisk shouldn't issue a reinvite because the yealinkphone is using ipv6 and the service provider, callwithus, is using ipv4. However, as I stated, I have directmedia=no, so nothing else should matter, but the statement is being ignored.
> I have worked around the issue by setting ,,t at the end of all of the Dial statements, and that successfully prevents switching to native_rtp.  Nothing else I tried does.



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