[asterisk-bugs] [JIRA] (ASTERISK-20674) nat=force_rport, comedia does not behave the same as nat=yes

[Alexandr Gordeev (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-20674?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=204396#comment-204396 ] 

Alexandr Gordeev commented on ASTERISK-20674:
---------------------------------------------

{code:xml}
<--- SIP read from UDP:95.138.160.45:6789 --->
OPTIONS sip:88.80.1.50 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.33:5060;branch=z9hG4bK7234c042;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 172.16.0.33>;tag=as7a4f4a22
To: <sip:88.80.1.50>
Contact: <sip:asterisk at 172.16.0.33:5060>
Call-ID: 3a181af619cfeb742df8da847babca49 at 172.16.0.33:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.15.0
Date: Tue, 19 Mar 2013 05:52:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<--- Transmitting (NAT) to 95.138.160.45:6789 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.33:5060;branch=z9hG4bK7234c042;received=95.138.160.45;rport=6789
From: "asterisk" <sip:asterisk at 172.16.0.33>;tag=as7a4f4a22
To: <sip:88.80.1.50>;tag=as12d5101b
Call-ID: 3a181af619cfeb742df8da847babca49 at 172.16.0.33:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 11.2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:88.80.1.50:5060>
Accept: application/sdp
Content-Length: 0
{code} 
                
> nat=force_rport,comedia does not behave the same as nat=yes
> -----------------------------------------------------------
>
>                 Key: ASTERISK-20674
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20674
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Core/Configuration
>    Affects Versions: 11.0.1
>         Environment: Gentoo Linux 64 bit PowerPC, kernel 3.6.1, GCC 4.5.3
>            Reporter: headphones
>            Assignee: Kinsey Moore
>            Severity: Minor
>         Attachments: ASTERISK-20674_debug.diff, Asterisk-debug-1-nat-yes-outgoing.log, Asterisk-debug-2-nat-yes-incoming.log, Asterisk-debug-3-nat-forcerportcomedia-outgoing.log, Asterisk-debug-4-nat-forcerportcomedia-incoming.log
>
>
> After an upgrade from 1.8.17.0 to 11.0.1 we get warnings about the deprecation of nat=yes in sip.conf:
> {{\[Nov 10 14:02:14\] WARNING\[28487\]: sip/config_parser.c:812 sip_parse_nat_option: nat=yes is deprecated, use nat=force_rport,comedia instead}}
> If we change to nat=force_rport,comedia the behavior seems to be fine, except for outside users behind NAT. Our server is also behind NAT.
> When an outside NAT:ed user calls in to the network everything works as expected, but when calling the outside user, or when two outside NAT:ed users call each other, the audio only goes one way without any errors shown in the console. When changing back to nat=yes, everything work again.
> By looking in the source code for sip/config_parser.c, it seems like the reason for the difference is that when using sip=yes, the auto_force_rport and auto_comedia are cleared from &flags\[2\]. However, when using force_rport,comedia, the auto equivalents are not cleared away.

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