[asterisk-bugs] [JIRA] (ASTERISK-20674) nat=force_rport, comedia does not behave the same as nat=yes
Alexandr Gordeev (JIRA)
noreply at issues.asterisk.org
Wed Mar 20 01:13:02 CDT 2013
[ https://issues.asterisk.org/jira/browse/ASTERISK-20674?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=204428#comment-204428 ]
Alexandr Gordeev commented on ASTERISK-20674:
---------------------------------------------
{quote}what are your NAT settings ?{quote}
I'm setting NAT parameter only in [General] section.
{code}asterisk -rx "sip show settings"
Global Settings:
----------------
UDP Bindaddress: 88.80.1.50:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: Yes
Allow unknown access: No
Allow subscriptions: No
Allow overlap dialing: No
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: Asterisk PBX 11.2.0
SDP Session Name: Asterisk PBX 11.2.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: asterisk
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: Yes
T.38 EC mode: Redundancy
T.38 MaxDtgrm: -1
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No
Network QoS Settings:
---------------------------
IP ToS SIP: CS0
IP ToS RTP audio: CS0
IP ToS RTP video: CS0
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No
Network Settings:
---------------------------
SIP address remapping: Disabled, no localnet list
Externhost: <none>
Externaddr: (null)
Externrefresh: 10
Global Signalling Settings:
---------------------------
Codecs: (gsm|ulaw|alaw|h263|testlaw)
Codec Order: none
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 60
RTP Hold Timeout: 120
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: No
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 900 secs
Reg. default duration: 600 secs
Sub. min duration 60 secs
Sub. max duration: 900 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Include CID: No
Notify hold state: No
SIP Transfer mode: closed
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy: <not set>
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70
Default Settings:
-----------------
Allowed transports: UDP
Outbound transport: UDP
Context: default
Record on feature: automon
Record off feature: automon
Force rport: Auto (No)
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: Never
Language: ru
Tone zone: ru
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk
----
Asterisk ending (0).{code}
{code}asterisk -rx "sip show peer office"
* Name : office
Description :
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : from-office
Record On feature : automon
Record Off feature : automon
Subscr.Cont. : <Not set>
Language : ru
Tonezone : ru
AMA flags : Unknown
Transfer mode: closed
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Named Callgr :
Nam. Pickupgr:
MOH Suggest :
Mailbox :
VM Extension : asterisk
LastMsgsSent : 0/0
Call limit : 50
Max forwards : 0
Dynamic : No
Callerid : "" <>
MaxCallBR : 384 kbps
Expire : -1
Insecure : no
Force rport : Auto (No)
Symmetric RTP: Yes
ACL : Yes
DirectMedACL : No
T.38 support : Yes
T.38 EC mode : Redundancy
T.38 MaxDtgrm: -1
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
Subscriptions: No
Overlap dial : No
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost : 95.138.160.45
Addr->IP : 95.138.160.45:6789
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: office
SIP Options : replaces replace timer
Codecs : (g729)
Codec Order : (g729:20)
Auto-Framing : No
Status : OK (3 ms)
Useragent :
Reg. Contact :
Qualify Freq : 60000 ms
Keepalive : 0 ms
Variables :
FAXOPT(gateway) = yes
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No
Asterisk ending (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: Alexandr Gordeev
> 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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