[asterisk-users] Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error

[Administrator TOOTAI]

Hi Matthew

Le 28/05/2012 19:28, Matthew J. Roth a écrit :
> Administrator TOOTAI wrote:
>
>> we are upgrading our Asterisk production server from 1.6.24 to 1.8.12
>> version and face the following problem: one of our peer
>> (voicetrading.com) doesn't accept our calls anymore, we receive a
>> timeout error "Packet timed out after 32000ms with no response".
>>
>> Switching back to 1.6 make things working again!
>>
>> In sip.conf we have nat=no, peer conf is:
>
> Asterisk 1.8.12 is not getting responses to the INVITES it sends.
> Comparing the INVITES, the only significant difference I see is that
> Asterisk 1.6.24 includes the "rport" field in the Via header and
> Asterisk 1.8.12 does not:
>
>    1.6.24 - Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK58aef527;rport
>    1.8.12 - Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK0c8907be
>
> Try setting "nat=force_rport" in sip.conf.  Please reply back to the
> list with the results.

We tested this setting this WE, effectively this problem disappear but 
another appears: call get connected but no audio. We installed Asterisk 
10.3.1 -> connection and no audio too, so same behaviour.

>
> There may be other differences between the versions that you haven't
> accounted for.  Read the CHANGES and UPGRADE.txt files in the root of
> the Asterisk source tree for details.

We did read those files, don't see which parameter we could have forget. 
media_address nor nat=comedia seems options for us. Hereunder a debug 
from call with force_rport: as you can see, the RTP audio is coming from 
another IP (77.77.777.77) We think asterisk doesn't accept this and 
don't know which parameter could solve this.


<--- SIP read from UDP:111.111.1.111:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK04e390b0;rport
From: "TOOTAi" <sip:0033333333333 at 222.222.22.22>;tag=as1335adb1
To: <sip:0000033666666666 at 111.111.1.111>;tag=4e0313ac670313ac4f9920c31847cea
Contact: sip:0000033666666666 at 111.111.1.111:5060
Call-ID: 72d5d3df06c07cc6037786ee59f574df at 222.222.22.22:5060
CSeq: 102 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
Content-Length: 159

v=0
o=CARRIER 1338276550 1338276550 IN IP4 77.77.777.77
s=SIP Call
c=IN IP4 77.77.777.77
t=0 0
m=audio 41462 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
<------------->
--- (11 headers 8 lines) ---
Found RTP audio format 0
Found audio description format PCMU for ID 0
Capabilities: us - (ulaw|alaw), peer - 
audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), 
combined - 0x0 (nothing)
Peer audio RTP is at port 77.77.777.77:41462
list_route: hop: <sip:0000033666666666 at 111.111.1.111:5060>
set_destination: Parsing <sip:0000033666666666 at 111.111.1.111:5060> for 
address/port to send to
set_destination: set destination to 111.111.1.111:5060
Transmitting (NAT) to 111.111.1.111:5060:
ACK sip:0000033666666666 at 111.111.1.111:5060 SIP/2.0
Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK0d106caa;rport
Max-Forwards: 70
From: "TOOTAi" <sip:0033333333333 at 222.222.22.22>;tag=as1335adb1
To: <sip:0000033666666666 at 111.111.1.111>;tag=4e0313ac670313ac4f9920c31847cea
Contact: <sip:0033333333333 at 222.222.22.22:5060>
Call-ID: 72d5d3df06c07cc6037786ee59f574df at 222.222.22.22:5060
CSeq: 102 ACK
User-Agent: TOOTAiAudio
Content-Length: 0


---
     -- SIP/myPeerDef-00000003 answered SIP/104-00000002

Thanks for your support.

-- 
Daniel