[asterisk-users] Critical - No audio issue with re-invite (wrong media address)

Hoa Thai Duy hoathai at vngt.vn
Mon Oct 16 21:36:43 MST 2006


Hi all

 

I faced the issue with canreinvite=yes in Asterisk (both 1.2.9.1 and
1.2.1.12).

 

My diagram

 

UA1(RFC1918) ----- ADSL1 (NAT) -------------- Asterisk (Public IP)
------------ (NAT)ADSL2 --- UA2 (RFC1918)

 

When canreinvite=no in sip.conf for both UAs, everything is fine, RTP from
both UAs are sent to Asterisk (can see using "rtp debug") and both UAs can
talk.

When canreinvite=yes and nat=yes, I faced no audio issue.

 

After debug, I found that, when Asterisk send re-invite to both UAs, both
UAs reply their OK with c=their RFC1918 address, and Asterisk send that OK
parameters to other UAs

 

Asterisk send re-invite to UA1 with c=Public NATed IP of ADSL2

Asterisk send re-invite to UA2 with c=Public NATed IP of ADSL1

UA1 reply OK to Asterisk with c=RFC1918 of UA1

UA2 reply OK to Asterisk with c=RFC1918 of UA2

Asterisk send OK to UA2 with c= RFC1918 of UA1 (because it received the
newer c= RFC1918 of UA1)

Asterisk send OK to UA1 with c= RFC1918 of UA2 (because it received the
newer c= RFC1918 of UA2)

 

UA1 send RTP directly to c= RFC1918 of UA2 (it must be c=Public NATed IP of
ADSL2 for correct audio flow)

UA2 send RTP directly to c= RFC1918 of UA1 (it must be c=Public NATed IP of
ADSL1 for correct audio flow)

. no audio at all .

 

 

Pls. help

 

Brgds

Hoa

 

Sip.conf

[general]

context=default

port=5060

bindaddr=0.0.0.0

realm=

srvlookup=no

disallow=all

allow=g729  

canreinvite=yes

nat=yes



 

[2222]

type=friend

host=dynamic

username=2222

secret=2222

canreinvite=yes         

nat=yes

callerid="2222" <2222>

allow=g729

 

[1111]

type=friend

host=dynamic

username=1111

secret=1111

canreinvite=yes         

nat=yes

callerid="1111" <1111>

allow=g729

 

 

extensions.conf

 

[general]

static=yes

writeprotect=no

exten => 2222,1,Answer

exten => 2222,2,Dial(SIP/2222)

exten => 2222,2,Hangup

 

exten => 1111,1,Answer

exten => 1111,2,Dial(SIP/1111)

exten => 1111,2,Hangup

 

 

 

 

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