Asterisk 1.4 and reinvite

Asterisk Asterisk Support
1

Hi

I don’t know why it happens, so, I’m here to ask you. I has installed Asterisk 1.4.0 and it works very well with the option CANREINVITE=UPDATE. The file sip.conf tells:

“a new feature in 1.4 - setting up the call directly between the endpoints instead of sending a re-INVITE”

The UPDATE is sent to each devices when one of them sends a BYE. But this behavior is VERY DIFFERENT in versions 1.4.1 and 1.4.2. Both works like version 1.2, sending a INVITE/UPDATE after 200 OK (the beginning of the RTP session). I want to use 1.4.2, but I need Asterisk do what it promises.

What is happing? Why this two versions don’t work like 1.4.0?

2

My error…

It is necessary to set in sip.conf:

directrtpsetup=yes

“This sets up the call directly with media peer-2-peer without re-invites”

3

Hi.
I’ve been trying the new experimental directrtpsetup feature but I cannot make it work. My sip.conf and extension.conf are very simple and the codec/fmtp for the call is the same for caller and callee.

[color=green]
SIP.CONF

[ul]
[general]
port = 5060
bindaddr = 0.0.0.0
allowguest=no
allowtransfer=no
;progressinband=yes
sipdebug = yes
recordhistory=yes
dumphistory=yes
canreinvite=no
directrtpsetup=yes
srvlookup=yes
useragent=B2BUA02
t38pt_udptl = yes
dtmfmode=rfc2833
disallow=all
allow=ulaw
rtptimeout=60
rtpholdtimeout=300
nat=never

[SIP_LCR]
type=peer
host=siplcr.mydomain.net
;insecure=port
dtmfmode=rfc2833
canreinvite=no
fromdomain=b2bua02.mydomain.net
rtptimeout=60
rtpholdtimeout=300
disallow=all
allow=ulaw

[EXT_LCR]
type=peer
host=sipext.mydomain.net
context=INC_EXT_LCR
rtptimeout=60
rtpholdtimeout=300
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=ulaw
[/ul]

EXTENSIONS.CONF

[ul]
[general]
static=yes

writeprotect=no

[default]

[INC_EXT_LCR]

exten => _1616#.,1,Dial(SIP/${EXTEN:5}@SIP_LCR)
exten => _1616#.,2,Hangup()

[/ul]
[/color]

As soon as Asterisk receive the invite, it sends a TRYING to caller and INVITE to callee but the SDP of it has media address of asterisk  :question: 
am I missing something here?, Thanks in advance.

[color=blue]
<— SIP read from 10.10.10.232:5060 —>
INVITE sip:1616#0212005622408171@10.10.10.247:5060 SIP/2.0
Record-Route: sip:10.10.10.232;ftag=6b466110a4;lr=on
Via: SIP/2.0/UDP 10.10.10.232;branch=z9hG4bK36c2.774825b6.0
Via: SIP/2.0/UDP 10.10.10.233:5060;branch=z9hG4bK6b466110a4126
From: sip:7772001004@sipext.mydomain.net;tag=6b466110a4
To: sip:1616#12345@sipext.mydomain.net
Call-ID: 6be43046-cdda-61a4-8310-0002a400248b@10.10.10.233
CSeq: 126 INVITE
Supported: timer, replaces
Min-SE: 1800
Date: Thu, 26 Apr 2007 17:42:03 GMT
User-Agent: AddPac SIP Gateway
Contact: sip:7772001004@10.10.10.233
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 239
Max-Forwards: 16

v=0
o=7772001004 1177609323 1177609323 IN IP4 10.10.10.233
s=AddPac Gateway SDP
c=IN IP4 10.10.10.233
t=1177609323 0
m=audio 24366 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

<------------->
— (18 headers 10 lines) —
Sending to 10.10.10.232 : 5060 (no NAT)
Using INVITE request as basis request - 6be43046-cdda-61a4-8310-0002a400248b@10.10.10.233
Found peer 'EXT_LCR’
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 10.10.10.233:24366
Found description format PCMU for ID 0
Found description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.10.10.233:24366
Looking for 1616#0212005622408171 in INC_EXT_LCR (domain 10.10.10.247)
list_route: hop: sip:10.10.10.232;ftag=6b466110a4;lr=on

<— Transmitting (no NAT) to 10.10.10.232:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.232;branch=z9hG4bK36c2.774825b6.0;received=10.10.10.232
Via: SIP/2.0/UDP 10.10.10.233:5060;branch=z9hG4bK6b466110a4126
From: sip:7772001004@sipext.mydomain.net;tag=6b466110a4
To: sip:1616#12345@sipext.mydomain.net
Call-ID: 6be43046-cdda-61a4-8310-0002a400248b@10.10.10.233
CSeq: 126 INVITE
User-Agent: VOISS B2BUA02
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:1616#0212005622408171@10.10.10.247
Content-Length: 0

<------------>
– Executing [1616#0212005622408171@INC_EXT_LCR:1] Dial(“SIP/sipext.mydomain.net-08894188”, “SIP/0212005622408171@SIP_LCR”) in new stack
Audio is at 10.10.10.247 port 17574
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.10.10.241:5060:
INVITE sip:0212005622408171@10.10.10.241 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.247:5060;branch=z9hG4bK61e54a36
From: “7772001004” sip:7772001004@b2bua02.mydomain.net;tag=as24b34282
To: sip:0212005622408171@10.10.10.241
Contact: sip:7772001004@10.10.10.247
Call-ID: 3832456d1cd0850b3bf8aacf77539cde@b2bua02.mydomain.net
CSeq: 102 INVITE
User-Agent: B2BUA02
Max-Forwards: 70
Date: Thu, 26 Apr 2007 17:40:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 26806 26806 IN IP4 10.10.10.247
s=session
c=IN IP4 10.10.10.247
t=0 0
m=audio 17574 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
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