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

Joshua Colp (JIRA) noreply at issues.asterisk.org
Thu Jan 19 05:29:10 CST 2017


     [ https://issues.asterisk.org/jira/browse/ASTERISK-26729?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Joshua Colp updated ASTERISK-26729:
-----------------------------------

    Assignee: Luke Escude
      Status: Waiting for Feedback  (was: Triage)

Have you tried using the "rtp_keepalive" option? This will send a keep alive RTP packet to ensure that the NAT mapping is opened and stays open even if no media is flowing.

Otherwise we'd need to see the console log with rtp set debug on and SIP traffic to see the precise flow.

> 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: Luke Escude
>
> 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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