[asterisk-bugs] [JIRA] Commented: (ASTERISK-20367) One-way audio with media_address

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-20367?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=197809#comment-197809 ] 

Rusty Newton commented on ASTERISK-20367:
-----------------------------------------

Richard, thanks for the clarification. From talking with developers here at Digium it's apparent that the described behavior for media_address is expected, but it doesn't mean that it is optimal. There is not a way currently within Asterisk to change bind address for RTP or the address that we send RTP from.

We would speculate that the phone is expecting to receive RTP only from the address it is also sending to for that session.

To change the behavior would technically be a feature request, unless there is a spec that we are not following somewhere or another reason to call it a bug.

>  One-way audio with media_address
> ---------------------------------
>
>                 Key: ASTERISK-20367
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20367
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 10.7.1
>            Reporter: Richard Kenner
>            Assignee: Rusty Newton
>
> I'm migrating from Asterisk 1.6.2 to 10.7.0.  In 1.6.2, I made a small patch to allow specifying an address for RTP media.  That worked.  In 10.7.0, this appears to be built in with "media_address", but it doesn't work for me.
> My Asterisk server has multiple addresses, all global address on two different /24's with different routing policies via BGP.  I'm connecting to a phone that's over NAT.  I have "nat=yes" in the "general" section of sip.conf.  Everything works fine with the default.
> But if I specify media_address to be the Asterisk server's address on the other /24, I get one-way audio.  I can see with "sip debug" that the proper address is being given in the SDP data.  Audio from the phone is fine. Audio *to* the phone starts out with maybe 1-2 seconds of very garbled audio, then goes quiet.
> Running traceroute shows that data comes from the phone *to* Asterisk on the desired /24, but goes out with a source address from the other /24 (the default address).  I'm not sure if this is the problem or not, but in any event, I think the source address for RTP should be the one in "media_address" and want it that way for my purposes anyway.  Is there a way to configure this to happen?  If not, where should I look to make a patch?  And is this likely the reason for the one-way audio or is something else the likely cause?

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