[asterisk-bugs] [JIRA] (ASTERISK-26729) Asterisk behind NAT not sending audio according to SDP

[Luke Escude (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26729?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=235816#comment-235816 ] 

Luke Escude commented on ASTERISK-26729:
----------------------------------------

Oh something interesting in that same debug log: ASterisk DOES start sending video, but to port 11256.... This port was never sent to Asterisk by kamailio at all. This port was in Asterisk's 200 OK.

If asterisk was properly handling NAT, it would not attempt to create a new port for video (11256) but should use Kamailio's assigned port (13452)

> Asterisk behind NAT not sending audio according to SDP
> ------------------------------------------------------
>
>                 Key: ASTERISK-26729
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26729
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 13.13.1
>         Environment: x64 CentOS
>            Reporter: Luke Escude
>            Assignee: Unassigned
>         Attachments: call1.pcap, Call1.txt, echo1.pcap, Echo1.txt, Video2.txt, video.pcap, Video.txt
>
>
> We are running Asterisk 13.13.1 with PJSIP.
> The setup is as follows:
> The Asterisk boxes are all virtualized behind NAT - let's say their address space is 172.x.x.x. The router that controls that nat is IP address 10.0.4.1. Kamailio/RTPProxy is running on 10.0.1.1.
> Kamailio essentially looks like a "public" IP to the asterisk boxes - we have them REGISTERING to kamailio to IP address 10.0.1.1 (which is routable of course). That way, Kamailio knows how to get to the Asterisks via their "public IP" (10.0.4.1) and NAT port (probably like 16875 or whatever).
> SIP communication happens wonderfully with this setup.
> However, when Kamailio sends an SDP containing its address 10.0.1.1 and an RTP address, it seems like Asterisk doesn't want to send audio to that address. In order for a new NAT port to open, Asterisk has to "talk" first.
> Regardless of our pjsip configuration, we cannot get Asterisk to behave properly, even by following the wiki and setting the external signaling/media addresses.



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