[asterisk-bugs] [Asterisk 0017851]: SIp/SDP for Non-NAT phone not used during 183 Session Progress with Non-NAT peer

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17851 
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Reported By:                whardier
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17851
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           Older 1.6.2 - please test a newer version 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-08-12 20:27 CDT
Last Modified:              2010-08-23 14:45 CDT
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Summary:                    SIp/SDP for Non-NAT phone not used during 183
Session Progress with Non-NAT peer
Description: 
SIP phone sends initial SDP for media path and makes a call through
chan_sip to a peer that sends back a 183 Session Progress with media.  I
cannot force the peer not to send a 183 it seems.

As soon as a 183 is received from the peer chan_sip sends a new SDP to the
phone and the peer to including itself in the media path and proxies the
media for the session progress between the peer and the phone.  Once
connected a new SDP is sent to the phone and the peer referencing eachother
and a native bridge is done.

Is this the intentional behavior of the SIP channel driver - will it
always attempt to proxy media when a 183 is present?  I am currently
attempting to save bandwidth by using the 'r' flag when dialing this peer
to keep from sending the phone session audio.  Session audio is still
present of course between the peer and Asterisk system.

I am classifying this as major priority since the difference in bandwidth
usage is substantial when you plan on only handling SIP traffic without
media.  The servers I am using handle media for some peers that don't have
heavy bandwidth requirements - however when you plan on only handling
signalling for some heavier peers the session progress can eat all your
bandwidth pretty quickly.
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---------------------------------------------------------------------- 
 (0126238) whardier (reporter) - 2010-08-23 14:45
 https://issues.asterisk.org/view.php?id=17851#c126238 
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At first I thought this was because I was using local channels.. I set up a
sip->sip dialplan to test this.. it seems as though if the phone sends an
sdp in the invite chan_sip sets up a 183 even if it's told to ignore sdp in
invites.  However.. the Asterisk box did not send a 183 to the provider
this time around since it didn't use local channels to route the call.

--- sip.conf
[general]
context=default
allowguest=no
match_auth_username=yes
allowoverlap=no

[vz-outbound](!)
type=peer
disallow=all
allow=g729
allow=ulaw
sendrpid=yes
canreinvite=yes
directmedia=yes

[vz-67.77.76.248](vz-outbound)
host=67.77.76.248
[vz-67.217.40.210](vz-outbound)
host=67.217.40.210

[libertytest]
type=friend
accountcode=9718210178
host=dynamic
allow=g729
secret=poot
canreinvite=yes
directmedia=yes
context=inbound
callerid=Liberty Telecom <9072640000>

--- extensions.conf
[general]
static=yes
writeprotect=no
clearglobalvars=no

[globals]

[inbound]
exten => 19078562879,1,Dial(SIP/vz-67.217.40.210/${EXTEN},1200)

--- Console

Connected to Asterisk 1.6.2.9-0.8541 currently running on sbc01 (pid =
21553)
Verbosity is at least 20
Core debug is at least 20
  == Using SIP RTP CoS mark 5
    -- Executing [19078562879 at inbound:1] Dial("SIP/libertytest-00000048",
"SIP/vz-67.217.40.210/19078562879,1200") in new stack
  == Using SIP RTP CoS mark 5
    -- Called vz-67.217.40.210/19078562879
    -- SIP/vz-67.217.40.210-00000049 is making progress passing it to
SIP/libertytest-00000048
    -- SIP/vz-67.217.40.210-00000049 answered SIP/libertytest-00000048
    -- Native bridging SIP/libertytest-00000048 and
SIP/vz-67.217.40.210-00000049
    -- Executing [h at inbound:1] Hangup("SIP/libertytest-00000048", "") in
new stack
  == Spawn extension (inbound, 19078562879, 1) exited non-zero on
'SIP/libertytest-00000048' 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-08-23 14:45 whardier       Note Added: 0126238                          
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