[asterisk-bugs] [JIRA] (ASTERISK-28993) PJSIP picks wrong media IP address for listening RTP

[Joshua C. Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-28993?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=251710#comment-251710 ] 

Joshua C. Colp commented on ASTERISK-28993:
-------------------------------------------

There is no ability that I'm aware of to read that kind of information. If it is present, it would be through the CHANNEL dialplan function like you've mentioned.

> PJSIP picks wrong media IP address for listening RTP
> ----------------------------------------------------
>
>                 Key: ASTERISK-28993
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28993
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip, Resources/res_pjsip_sdp_rtp
>    Affects Versions: 16.11.0
>         Environment: Debian GNU/Linux 9
>            Reporter: Marin Odrljin
>            Severity: Minor
>         Attachments: ari-app.log, full-log-filtered, http.conf, pjsip.conf, pjsip-new.conf, rtp.conf
>
>
> We are having multiple local IP addresses 10.5.20.42 ,.52, ,.62, ,.72 for multiple PJSIP trunks toward 2 different provider IP addresses. SIP INVITE sends SDP as following:
> {code}
> c=IN IP4 10.5.20.42
> m=audio 12442 RTP/AVP 8 3 101
> {code}
> but UDP listening address is the last one .72:
> {code}
> ss -na
> udp    UNCONN     0      0      10.5.20.72:12442                 *:*
> {code}
> So the result is no incoming RTP packets are comming into Asterisk - no IN audio.
> Intersting thing is that in Asterisk 13 we have had the same configuration and it worked because Asterisk was listening on all IPs 0.0.0.0



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