[asterisk-bugs] [JIRA] (ASTERISK-25676) native_rtp Lets through Codec not in the Allowed List instead of Transcoding

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25676?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=228907#comment-228907 ] 

Asterisk Team commented on ASTERISK-25676:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

> native_rtp Lets through Codec not in the Allowed List instead of Transcoding
> ----------------------------------------------------------------------------
>
>                 Key: ASTERISK-25676
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25676
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 13.7.0
>         Environment: Debian 8.2
>            Reporter: Peter Sokolov
>
> The call originating SIP device only has g722 in the list of allowed codecs and directmedia=no set in its configuration. On a call to SIP destination that replies with alaw (183 and OK), Asterisk 13 will correctly transcode alaw to g722 during the early media, however as soon as the connection is established (OK) and the bridge switches to native_rtp, it will simply let through the alaw packets.
> To reproduce, configure like below. 333 is a SIP phone (FritzBox in my case) that supports g722 and alaw. OutTrunk is a SIP trunk provider that does not support g722 for calls out of its network (it replies with alaw only in 183 and OK SDP). At least one peer has to be set to directmedia=no in a way that RTP always flows via Asterisk.
> Make a call from peer 333 to a number that is routed through OutTrunk. When OutTrunk sends 183 with only alaw in SDP, you will see that Asterisk sends 183 with g722 in SDP to peer 333 and will transcode the alaw early media to g722. As soon as the connection is established and OutTrunk sends OK with only alaw in SDP, you will see that Asterisk sends OK with g722 in SDP to peer 333, however native_rtp bridge will let through the alaw media to the peer 333 even though alaw is not in the list of allowed codecs for the peer 333.
> {code}
> sip.conf:
> [333]
> type=friend
> context=users
> subscribecontext=users
> secret=donttell
> language=de
> mailbox=333 at default
> host=dynamic
> dtmfmode=rfc2833
> defaultuser=333
> callerid=333 <00333>
> restrictcid=no
> fromdomain=test.tst
> directmedia=no
> nat=no
> insecure=port
> call-limit=5
> allowsubscribe=yes
> disallow=all
> allow=g722
> [OutTrunk]
> type=peer
> fromdomain=test2.tst
> defaultuser=user1
> secret=donttell
> host=test2.tst
> context=users
> directmedia=yes
> qualify=no
> insecure=port,invite
> promiscredir=no
> dtmfmode=rfc2833
> nat=no
> disallow=all
> allow=g722,alaw
> callgroup=1
> t38pt_udptl=no
> t38pt_rtp=no
> t38pt_tcp=no
> extensions.conf:
> [users]
> exten => 123,1,NoOp
> exten => 123,n,Set(CALLERID(number)=00333)
> exten => 123,n,Set(CALLERID(name)=${CALLERID(number):2})
> exten => 123,n,Set(CALLERID(number)=${CALLERID(number):2})
> exten => 123,n,Dial(SIP/${EXTEN}@OutTrunk,60,)
> exten => 123,n,Hangup
> {code}



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