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

[Marin Odrljin (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-28993?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Marin Odrljin updated ASTERISK-28993:
-------------------------------------

    Attachment: ari-app.log

I have attached my ARI app log file as well because I have found other strange issues related to PJSIP and reading out call data.

If you compare INVITE message SDP you'll see that local media address is set to 10.5.20.52:7366, but when app is reading 'CHANNEL(rtp,src,audio)' variable it gets back other value 10.5.20.42:7366 (ari-app.log line 24). Btw. 10.5.20.52 is correct one and audio works fine on that IP after setting 'media_address' into pjsip.conf, before that asterisk was using wrong IP.

A bit off topic, but our ARI app is trying to read some data on outbound PJSIP call via channel and pjsip functions, but unfortunatelly a lot of them doesn't work properly, e.g. reading CHANNEL(pjsip,remote_addr) or CHANNEL(pjsip,local_addr) returns 'Unable to read provided function'. Then PJSIP_HEADER(read,name) is available only on inbound channel. Is there any way to read out SIP headers such as From, To, Contact, Via etc. on outbound PJSIP call?

> 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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