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

[Richard Kenner (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-20367?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=197098#comment-197098 ] 

Richard Kenner commented on ASTERISK-20367:
-------------------------------------------

Let me try to clarify.  I'll call our primary network the "205 network" from the first octet.  The network that I want to use for RTP is the "207 network".  The interface's primary address is the 205 network.  All SIP data goes on that network.  With media_address set, the SDP entry contains the 207 network.  RTP packets from the phone come back on the 207 network and there's no problem with them.  Packets *to* the phone go out from the 205 network.  I'm not saying this is "wrong", but just that it's a possible explanation for the fact that, with the exception of the first second or so, the phone (an Aastra 6757i, which works fine without media_address) doesn't get them.

The relevant portions of sip.conf (all each phone has is fullname and md5secret) is:

[general]
nat=yes
qualify=yes
allowguest=no
alwaysauthreject=yes
realm=asterisk.gnat.com
sendrpid=yes
trustrpid=yes
videosupport=yes
usereqphone=yes
callcounter=yes
directmedia=no
qualifyfreq=240
defaultexpiry=600
;media_address=207.xx.xx.xx

[phones](!)
type=friend
host=dynamic
context=SIP_Phones
cc_agent_policy=generic
cc_monitor_policy=generic
disallow=all
allow=gsm
allow=ulaw
allow=g729

I can attach a packet trace, but is it still needed given the above?

>  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: Richard Kenner
>
> 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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