[asterisk-bugs] [JIRA] Issue Comment Edited: (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=197063#comment-197063 ] 

Rusty Newton edited comment on ASTERISK-20367 at 9/14/12 9:39 AM:
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{quote}
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)
{quote}
Do you mean that the SDP the phone sends contains a media address from the other /24 ? 

If I understand the media_address setting, it will only affect the media address for SDP outbound from Asterisk. It sounds like you want to affect the media address being sent *to* Asterisk by a phone. I may be failing to understand what you wrote.

Can you provide a sanitized sip.conf, plus a packet capture (a tcpdump pcap if possible) of the SIP and RTP traffic for all legs of the calls. We'd like to look at a bit closer to make sure it's expected behavior.


      was (Author: rnewton):
    {quote}
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)
{quote}
Do you mean that the SDP the phone sends contains a media address from the other /24 ? 

If I understand the media_address setting, it will only affect the media address for SDP outbound from Asterisk. It sounds like you want to affect the media address being sent *to* Asterisk by a phone. I may be failing to understand what you wrote.

Can you provide a sanitized sip.conf, plus a packet capture of the SIP and RTP traffic for all legs of the calls. We'd like to look at a bit closer to make sure it's expected behavior.

  
>  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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