[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:all-tabpanel ]

Luke Escude updated ASTERISK-26729:
-----------------------------------

    Attachment: echo1.pcap
                Echo1.txt
                call1.pcap
                Call1.txt

Attached two PCAPs and debug log outputs for a couple of example calls that don't have two-way audio.

I believe the INVITE coming from kamailio is receiving a TRYING response from Asterisk, but asterisk isn't actually doing anything.

> 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
>
>
> 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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