[asterisk-bugs] [JIRA] (ASTERISK-28901) pjsip behaves incorrectly when sending RTP, it sends it to a private IP

Asterisk Team (JIRA) noreply at issues.asterisk.org
Sun May 17 08:22:25 CDT 2020


    [ https://issues.asterisk.org/jira/browse/ASTERISK-28901?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=250799#comment-250799 ] 

Asterisk Team commented on ASTERISK-28901:
------------------------------------------

The severity of this issue has been automatically downgraded from "Blocker" to "Major". The "Blocker" severity is reserved for issues which have been determined to block the next release of Asterisk. This severity can only be set by privileged users. If this issue is deemed to block the next release it will be updated accordingly during the triage process.

> pjsip behaves incorrectly when sending RTP, it sends it to a private IP
> -----------------------------------------------------------------------
>
>                 Key: ASTERISK-28901
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28901
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip
>    Affects Versions: 13.33.0, 16.10.0
>         Environment: Linux
>            Reporter: Private Name
>
> I found that the PJSIP incorrectly sends RTP back to a private address, while in the same exact circumstances the old sip channel works correctly.
> My phone is located behind a NAT, 172.16.0.0/21.
> Asterisk 16 is on a public IP.
> PJSIP has the config below:
> force_rport=yes
> direct_media=yes
> disable_direct_media_on_nat = yes
> direct_media_method=invite
> But when I send a call I see the RTP being sent to my private address, vs the public IP. This only happens when Asterisk  has dialed the call to another carrier. If instead of Dial I choose Answer() and MusicOnHold, then the RTP gets shipped to the right address.
> This is a sample of the erroneous behavior:
> Got  RTP packet from    XX.XX.XX.XX:17510 (type 00, seq 024786, ts 017440, len 000160)
> Sent RTP packet to      172.16.7.254:50798 (type 00, seq 010736, ts 017440, len 000160)
> 172.16.7.254 is my private address
> This is the call flow:
> Sostphone --->Router---->Asterisk16Public--->Asterisk16Public (musicOnHold)
> On the first Asterisk16Public I see the RTP being sent to the Private IP.
> I am uploading my PJSIP.conf
> The dialplan is: dial(${EXTEN}@asterisk)



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