[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:comment-tabpanel&focusedCommentId=251443#comment-251443 ] 

Marin Odrljin commented on ASTERISK-28993:
------------------------------------------

Ok, thank you for your answer. Here is the INVITE message which shows that the same behaviour is there:
{code}
INVITE sip:+41786144341 at X.X.X.135 SIP/2.0
Via: SIP/2.0/UDP 10.5.20.42:5060;rport;branch=z9hG4bKPj199eb8e5-b0a8-4cda-b156-71f295492c45
From: <sip:+41413691000 at 10.5.20.72>;tag=8b2bd8e7-e2b1-4cd4-b265-4d971ba82290
To: <sip:+41786144341 at X.X.X.135>
Contact: <sip:asterisk at 10.5.20.42:5060>
Call-ID: f21caeaa-e059-4199-b4e2-d25053152f8e
CSeq: 6931 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 360
Max-Forwards: 70
User-Agent: Asterisk PBX 16.11.0
Content-Type: application/sdp
Content-Length:   256

v=0
o=- 1308455539 1308455539 IN IP4 10.5.20.72
s=Asterisk
c=IN IP4 10.5.20.72
t=0 0
m=audio 12442 RTP/AVP 8 3 101
...
{code}

This is a call that has transport and media address .72. In Via and Contact headers there is .42 while in From and SDP it is .72 (with media_address settings also now corrected in SDP). Luckily our ISP allows that because this is a kind of LAN traffic with them on all of those IPs. Different IPs are used for different trunks toward them for different clients. But I'm afraid that this is not completely correct - probably signalling should also go through the same corresponding transport IP.

> 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
>            Assignee: Unassigned
>            Severity: Minor
>         Attachments: http.conf, pjsip.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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