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Summary:ASTERISK-19050: Wrong transport for outgoing INVITE
Reporter:Dmitry Alekseev (rara avis)Labels:
Date Opened:2011-12-15 04:35:58.000-0600Date Closed:2018-01-02 08:44:24.000-0600
Priority:MajorRegression?Yes
Status:Closed/CompleteComponents:Channels/chan_sip/Messaging Channels/chan_sip/TCP-TLS
Versions:1.8.7.2 Frequency of
Occurrence		Constant
Related
Issues:
Environment:Minimal CentOS 5.7, Asterisk 1.8.7.0 from repo, updated to 1.8.7.2, behind the NAT.Attachments:
Description:INVITE is sent via UDP despite previous REGISTER via TCP and TCP being declared as primary outboud transport.

REGISTERing to external provider via TCP - Ok.
But outgoing INVITE to the same provider is sent via UDP, despite TCP being configured primary outgoing transport, 403 Forbidden as result.
All combinations of "transport=tcp" or "transport=tcp,udp" in general or peer section or in both - behave the same.
1.8.7.0 - the same.

1.6.2.21 - working as expected (INVITE via TCP)



sip.conf:

{noformat}
[general]
context=default
allowguest=no
allowoverlap=no
alwaysauthreject=yes
udpbindaddr = 0.0.0.0

tcpenable = yes
tcpbindaddr = 0.0.0.0
transport=tcp,udp

srvlookup=yes
nat = yes
disallow=all
allow=alaw
allow=g729
allow=g723
allow=ulaw
dtmfmode=inband
canreinvite=no
externip=AAA.BBB.CCC.DDD
register = tcp://7AAAAAAAAAA@multifon.ru:password:7AAAAAAAAAA@sbc.megafon.ru/7AAAAAAAAAA

[2provider]
type=peer
host = multifon.ru
username = 7AAAAAAAAAA
secret = password
context = default
outboundproxy = sbc.megafon.ru
fromdomain = multifon.ru
fromuser = 7AAAAAAAAAA
authuser = 7AAAAAAAAAA
insecure = port,invite
{noformat}

extensions.conf:

{noformat}
[general]
static = yes
writeprotect = no
clearglobalvars = no

[default]
exten=>_7XXXXXXXXXX,1,Dial(SIP/${EXTEN}@2provider,30,r)
{noformat}

show's:

{noformat}
localhost*CLI> core show version
Asterisk 1.8.7.2 built by root @ localhost.localdomain on a i686 running Linux on 2011-12-09 17:52:28 UTC
localhost*CLI>


localhost*CLI> sip show settings


Global Settings:
----------------
 UDP Bindaddress:        0.0.0.0:5060
 TCP SIP Bindaddress:    0.0.0.0:5060
 TLS SIP Bindaddress:    Disabled
 Videosupport:           No
 Textsupport:            No
 Ignore SDP sess. ver.:  No
 AutoCreate Peer:        No
 Match Auth Username:    No
 Allow unknown access:   No
 Allow subscriptions:    Yes
 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 1.8.7.2
 SDP Session Name:       Asterisk PBX 1.8.7.2
 SDP Owner Name:         root
 Reg. context:           (not set)
 Regexten on Qualify:    No
 Legacy userfield parse: No
 Caller ID:              asterisk
 From: Domain:
 Record SIP history:     Off
 Call Events:            Off
 Auth. Failure Events:   Off
 T.38 support:           No
 T.38 EC mode:           Unknown
 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:             AAA.BBB.CCC.DDD:0
 Externrefresh:          10

Global Signalling Settings:
---------------------------
 Codecs:                 0x10d (g723|ulaw|alaw|g729)
 Codec Order:            alaw:20,g729:20,g723:30,ulaw:20
 Relax DTMF:             No
 RFC2833 Compensation:   No
 Symmetric RTP:          Yes
 Compact SIP headers:    No
 RTP Keepalive:          0 (Disabled)
 RTP Timeout:            0 (Disabled)
 RTP Hold Timeout:       0 (Disabled)
 MWI NOTIFY mime type:   application/simple-message-summary
 DNS SRV lookup:         Yes
 Pedantic SIP support:   Yes
 Reg. min duration       60 secs
 Reg. max duration:      3600 secs
 Reg. default duration:  120 secs
 Outbound reg. timeout:  20 secs
 Outbound reg. attempts: 0
 Notify ringing state:   Yes
   Include CID:          No
 Notify hold state:      No
 SIP Transfer mode:      open
 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:     TCP,UDP
 Outbound transport:     TCP
 Context:                default
 Force rport:            Yes
 DTMF:                   inband
 Qualify:                0
 Use ClientCode:         No
 Progress inband:        Never
 Language:
 MOH Interpret:          default
 MOH Suggest:
 Voice Mail Extension:   asterisk

----
localhost*CLI> sip show peer 2provider


 * Name       : 2provider
 Secret       : <Set>
 MD5Secret    : <Not set>
 Remote Secret: <Not set>
 Context      : default
 Subscr.Cont. : <Not set>
 Language     :
 AMA flags    : Unknown
 Transfer mode: open
 CallingPres  : Presentation Allowed, Not Screened
 FromUser     : 7AAAAAAAAAA
 FromDomain   : multifon.ru Port 5060
 Callgroup    :
 Pickupgroup  :
 MOH Suggest  :
 Mailbox      :
 VM Extension : asterisk
 LastMsgsSent : 32767/65535
 Call limit   : 0
 Max forwards : 0
 Dynamic      : No
 Callerid     : "" <>
 MaxCallBR    : 384 kbps
 Expire       : -1
 Insecure     : port,invite
 Force rport  : Yes
 ACL          : No
 DirectMedACL : No
 T.38 support : No
 T.38 EC mode : Unknown
 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: Yes
 Overlap dial : No
 Outb. proxy  : sbc.megafon.ru
 DTMFmode     : inband
 Timer T1     : 500
 Timer B      : 32000
 ToHost       : multifon.ru
 Addr->IP     : 193.201.229.35:5060
 Defaddr->IP  : (null)
 Prim.Transp. : TCP
 Allowed.Trsp : TCP,UDP
 Def. Username: 7AAAAAAAAAA
 SIP Options  : (none)
 Codecs       : 0x10d (g723|ulaw|alaw|g729)
 Codec Order  : (alaw:20,g729:20,g723:30,ulaw:20)
 Auto-Framing :  No
 100 on REG   : No
 Status       : Unmonitored
 Useragent    :
 Reg. Contact :
 Qualify Freq : 60000 ms
 Sess-Timers  : Accept
 Sess-Refresh : uas
 Sess-Expires : 1800 secs
 Min-Sess     : 90 secs
 RTP Engine   : asterisk
 Parkinglot   :
 Use Reason   : No
 Encryption   : No

localhost*CLI>
{noformat}

debug:

{noformat}
localhost*CLI> sip set debug peer 2provider

localhost*CLI>
SIP Debugging Enabled for IP: 193.201.229.35

localhost*CLI>
[Dec 14 16:20:19] NOTICE[3535]: chan_sip.c:12596 sip_reregister:    -- Re-registration for  7AAAAAAAAAA@sbc.megafon.ru

localhost*CLI>
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 193.201.229.35:5060:
REGISTER sip:multifon.ru SIP/2.0

Via: SIP/2.0/TCP 10.128.1.225:5060;branch=z9hG4bK3cedc749;rport

Max-Forwards: 70

From: <sip:7AAAAAAAAAA@multifon.ru>;tag=as176e1f32

To: <sip:7AAAAAAAAAA@multifon.ru>

Call-ID: 20861aac7e7569e62c3608ff7d62cd9e@127.0.0.1

CSeq: 105 REGISTER

User-Agent: Asterisk PBX 1.8.7.2

Authorization: Digest username="7AAAAAAAAAA", realm="BREDBAND", algorithm=MD5, uri="sip:multifon.ru", nonce="MTMyMzgzNjIzNDpfLjfk775Bd2kMNs4bxIr3", response="2fdc731e5f472d43843edc9d95b8ea9e", opaque="MTMyMzgzNjIzNDpfLjfk775Bd2kMNs4bxIr3", qop=auth, cnonce="74cc256c", nc=00000003

Expires: 120

Contact: <sip:7AAAAAAAAAA@10.128.1.225:5060;transport=TCP>

Content-Length: 0




---

localhost*CLI>

<--- SIP read from TCP:193.201.229.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.128.1.225:5060;received=AAA.BBB.CCC.DDD;branch=z9hG4bK3cedc749;rport=36591
From: <sip:7AAAAAAAAAA@multifon.ru>;tag=as176e1f32
To: <sip:7AAAAAAAAAA@multifon.ru>;tag=2CB932463135364126DE1600
Call-ID: 20861aac7e7569e62c3608ff7d62cd9e@127.0.0.1
CSeq: 105 REGISTER
Contact: <sip:7AAAAAAAAAA@10.128.1.225:5060;transport=TCP>;expires=130
Content-Length: 0
Service-Route: <sip:7AAAAAAAAAA@193.201.229.35:5060;transport=tcp;lr>

<------------->
--- (9 headers 0 lines) ---
Scheduling destruction of SIP dialog '20861aac7e7569e62c3608ff7d62cd9e@127.0.0.1' in 32000 ms (Method: REGISTER)
[Dec 14 16:20:19] NOTICE[3547]: chan_sip.c:20155 handle_response_register: Outbound Registration: Expiry for sbc.megafon.ru is 130 sec (Scheduling reregistration in 115 s)

localhost*CLI> console dial 7BBBBBBBBBB

localhost*CLI>
Audio is at 5060

localhost*CLI>
Adding codec 0x8 (alaw) to SDP

localhost*CLI>
Adding codec 0x4 (ulaw) to SDP

localhost*CLI>
Reliably Transmitting (NAT) to 193.201.229.35:5060:
INVITE sip:7BBBBBBBBBB@multifon.ru SIP/2.0

Via: SIP/2.0/UDP 10.128.1.225:5060;branch=z9hG4bK2d2b39d7;rport

Max-Forwards: 70

From: "asterisk" <sip:7AAAAAAAAAA@multifon.ru>;tag=as653efae6

To: <sip:7BBBBBBBBBB@multifon.ru>

Contact: <sip:7AAAAAAAAAA@10.128.1.225:5060>

Call-ID: 21bd2e403656248b19d83565196334ea@multifon.ru

CSeq: 102 INVITE

User-Agent: Asterisk PBX 1.8.7.2

Date: Wed, 14 Dec 2011 10:20:43 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 202



v=0

o=root 518834463 518834463 IN IP4 10.128.1.225

s=Asterisk PBX 1.8.7.2

c=IN IP4 10.128.1.225

t=0 0

m=audio 16650 RTP/AVP 8 0

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=ptime:20

a=sendrecv


---

localhost*CLI>

<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.128.1.225:5060;received=AAA.BBB.CCC.DDD;branch=z9hG4bK2d2b39d7;rport=1027
From: "asterisk" <sip:7AAAAAAAAAA@multifon.ru>;tag=as653efae6
To: <sip:7BBBBBBBBBB@multifon.ru>
Call-ID: 21bd2e403656248b19d83565196334ea@multifon.ru
CSeq: 102 INVITE

<------------->
--- (6 headers 0 lines) ---

<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.128.1.225:5060;received=AAA.BBB.CCC.DDD;branch=z9hG4bK2d2b39d7;rport=1027
From: "asterisk" <sip:7AAAAAAAAAA@multifon.ru>;tag=as653efae6
To: <sip:7BBBBBBBBBB@multifon.ru>;tag=aprqngfrt-k04r2m20000c6
Call-ID: 21bd2e403656248b19d83565196334ea@multifon.ru
CSeq: 102 INVITE
Reason: Q.850;cause=55;text="Call Terminated"

<------------->
--- (7 headers 0 lines) ---
Transmitting (NAT) to 193.201.229.35:5060:
ACK sip:7BBBBBBBBBB@multifon.ru SIP/2.0

Via: SIP/2.0/UDP 10.128.1.225:5060;branch=z9hG4bK2d2b39d7;rport

Max-Forwards: 70

From: "asterisk" <sip:7AAAAAAAAAA@multifon.ru>;tag=as653efae6

To: <sip:7BBBBBBBBBB@multifon.ru>;tag=aprqngfrt-k04r2m20000c6

Contact: <sip:7AAAAAAAAAA@10.128.1.225:5060>

Call-ID: 21bd2e403656248b19d83565196334ea@multifon.ru

CSeq: 102 ACK

User-Agent: Asterisk PBX 1.8.7.2

Content-Length: 0




---

localhost*CLI>
[Dec 14 16:20:43] WARNING[3535]: chan_sip.c:19680 handle_response_invite: Received response: "Forbidden" from '"asterisk" <sip:7AAAAAAAAAA@multifon.ru>;tag=as653efae6'

localhost*CLI>
Really destroying SIP dialog '21bd2e403656248b19d83565196334ea@multifon.ru' Method: INVITE

localhost*CLI>
Really destroying SIP dialog '20861aac7e7569e62c3608ff7d62cd9e@127.0.0.1' Method: OPTIONS

localhost*CLI>
Really destroying SIP dialog '20861aac7e7569e62c3608ff7d62cd9e@127.0.0.1' Method: REGISTER

localhost*CLI>
<< Hangup on console >>

localhost*CLI>

{noformat}


Comments:By: Dmitry Alekseev (rara avis) 2011-12-16 00:15:05.421-0600

Workaround:

outboundproxy = tcp://...

By: Luke H (luckman212) 2012-01-05 10:49:29.821-0600

Is this fixed as of the 1.8.8.1 or 1.8.9.0-rc1 ? Just curious as I may be having an issue related to this.
By: Dmitry Alekseev (rara avis) 2012-01-05 12:39:48.294-0600

The problem still persist in 1.8.8.1 .

By: Joshua C. Colp (jcolp) 2017-12-19 06:24:35.930-0600

Are you still experiencing this in chan_sip as of a currently supported version of Asterisk?
By: Asterisk Team (asteriskteam) 2018-01-02 08:44:24.369-0600

Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1].

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines