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[asnoc at teliax.com]

For some reason when trunking with Asterisk PBX's (yes, I know) FS
wants to send the ACK to the internal ip found in the Contact field of
the 200 OK. We have the force rport setting on but it's still not
responding to that IP. Register's work. Most of the sip signalling
works, just when the customer specifies the Contact filed with an
internal ip. Below is a packet capture and our external.xml conf file.

U 2008/11/04 09:17:08.259672 64.74.188.23:5060 -> 68.188.189.202:5060
INVITE sip:989607XXXX@68.188.189.202:5060 SIP/2.0.
Via: SIP/2.0/UDP 64.74.188.23;rport;branch=z9hG4bKc6y6pr80HyeaN.
Max-Forwards: 68.
From: "user" <sip:303452XXXX@64.74.188.23>;tag=Dt6v81cDZXa3B.
To: <sip:989607XXXX@68.188.189.202:5060>.
Call-ID: de7c471c-252e-122c-3cba-5f1bac93b621.
CSeq: 106789378 INVITE.
Contact: <sip:mod_sofia@64.74.188.23:5060>.
User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
Supported: 100rel, timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, call-info, sla, include-session-
description, presence.winfo, message-summary.
Min-SE: 120.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 370.

U 2008/11/04 09:21:32.861720 68.188.189.202:5060 -> 64.74.188.23:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP
64.74.188.23
;branch=z9hG4bKZFem0FmX4g19H;received=64.74.188.23;rport=5060.
From: "user" <sip:303452XXXX@64.74.188.23>;tag=4UF788r8ct8aD.
To: <sip:989607XXXX@68.188.189.202:5060>.
Call-ID: 7c2d8578-252f-122c-3cba-5f1bac93b621.
CSeq: 106789510 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
Contact: <sip:989607XXXX@192.168.0.5>.
Content-Length: 0.
.

U 2008/11/04 09:21:32.861845 68.188.189.202:5060 -> 64.74.188.23:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
64.74.188.23
;branch=z9hG4bKZFem0FmX4g19H;received=64.74.188.23;rport=5060.
From: "user" <sip:303452XXXX@64.74.188.23>;tag=4UF788r8ct8aD.
To: <sip:989607XXXX@68.188.189.202:5060>;tag=as1da4b7aa.
Call-ID: 7c2d8578-252f-122c-3cba-5f1bac93b621.
CSeq: 106789510 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
Contact: <sip:989607XXXX@192.168.0.5>.
Content-Type: application/sdp.
Content-Length: 285.
.
v=0.
o=root 10970 10970 IN IP4 192.168.0.5.
s=session.
c=IN IP4 192.168.0.5.
t=0 0.
m=audio 15876 RTP/AVP 18 0 101.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.


U 2008/11/04 09:21:32.862573 64.74.188.23:5060 -> 192.168.0.5:5060
ACK sip:989607XXXX@192.168.0.5 SIP/2.0.
Via: SIP/2.0/UDP 64.74.188.23;rport;branch=z9hG4bK0r7c2a501SQvD.
Max-Forwards: 70.
From: "user" <sip:303452XXXX@64.74.188.23>;tag=4UF788r8ct8aD.
To: <sip:989607XXXX@68.188.189.202:5060>;tag=as1da4b7aa.
Call-ID: 7c2d8578-252f-122c-3cba-5f1bac93b621.
CSeq: 106789510 ACK.
Contact: <sip:mod_sofia@64.74.188.23:5060>.
Content-Length: 0.
.





external.xml


<settings>
<param name="debug" value="0"/>
<param name="sip-trace" value="no"/>
<param name="rfc2833-pt" value="101"/>
<param name="sip-port" value="5060"/>
<param name="dialplan" value="XML"/>
<param name="context" value="public"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs" value="$${outbound_codec_prefs}"/>
<param name="hold-music" value="$${hold_music}"/>
<param name="use-rtp-timer" value="true"/>
<param name="rtp-timer-name" value="soft"/>
<param name="multiple-registrations" value="true"/>
<param name="manage-presence" value="true"/>
<param name="aggressive-nat-detection" value="true"/>
<param name="NDLB-force-rport" value="true"/>
<param name="inbound-codec-negotiation" value="generous"/>
<param name="nonce-ttl" value="60"/>
<param name="auth-calls" value="true"/>
<param name="rtp-timeout-sec" value="1800"/>
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="sip-ip" value="$${local_ip_v4}"/>
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
<param name="ext-sip-ip" value="$${external_sip_ip}"/>
<param name="rtp-timeout-sec" value="300"/>
<param name="rtp-hold-timeout-sec" value="1800"/>
</settings>

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[brian at freeswitch.org]

You need to set localnet and externip or externhost on Asterisk so it
doesn't lie about its IP.

/b

On Nov 4, 2008, at 11:10 AM, David Aldworth wrote:

[daldworth at teliax.com]

Hi bkw -

We did that and it does indeed fix the issue. However, in the case
that you have multiple SIP UA's behind a router there tend to be many
dynamically generated ports in use. The obvious solution would be to
statically map a port to an internal IP and then set the externip and
localhost settings. I agree, this would work. Except if you are using
a dsl or cable modem provider that also like to update your WAN ip on
a regular basis. But what confuses me more is that all the sip
messaging works fine right up until the ACK we send back to the 200
OK. Obviously FS is sending the ACK to the Contact field IP but is
there a way in FS to tell it to just respond on the IP and port that
the 200 OK came from? I thought that is what the force rport setting
did but i guess it does not.

David

On Nov 4, 2008, at 10:32 AM, Brian West wrote:

[brian at freeswitch.org]

You can use the param "NDLB-force-rport" to force it to use rport no
matter what.

/b

On Nov 4, 2008, at 11:43 AM, David Aldworth wrote:

[daldworth at teliax.com]

That is actually already on.

Any idea?

On Nov 4, 2008, at 10:45 AM, Brian West wrote:

[daldworth at teliax.com]

It was in there when the server got brought up.

Here is the full .xml settings:

<settings>
<param name="debug" value="0"/>
<param name="sip-trace" value="no"/>
<param name="rfc2833-pt" value="101"/>
<param name="sip-port" value="5060"/>
<param name="dialplan" value="XML"/>
<param name="context" value="public"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs" value="$${outbound_codec_prefs}"/>
<param name="hold-music" value="$${hold_music}"/>
<param name="use-rtp-timer" value="true"/>
<param name="rtp-timer-name" value="soft"/>
<param name="multiple-registrations" value="true"/>
<param name="manage-presence" value="true"/>
<param name="aggressive-nat-detection" value="true"/>
<param name="NDLB-force-rport" value="true"/>
<param name="inbound-codec-negotiation" value="generous"/>
<param name="nonce-ttl" value="60"/>
<param name="auth-calls" value="true"/>
<param name="rtp-timeout-sec" value="1800"/>
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="sip-ip" value="$${local_ip_v4}"/>
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
<param name="ext-sip-ip" value="$${external_sip_ip}"/>
<param name="rtp-timeout-sec" value="300"/>
<param name="rtp-hold-timeout-sec" value="1800"/>
</settings>


On Nov 4, 2008, at 10:52 AM, Brian West wrote:

[brian at freeswitch.org]

Make sure you restart the profile for it to take effect.

/b

On Nov 4, 2008, at 11:49 AM, David Aldworth wrote:

[anthony.minessale at g...]

keep in mind that the spec says when a contact addr has a new ip that you are supposed to
change the ip for all the rest of the messages to that new location. So we are doing what we are supposed to.

If you want to force the behavior you can use the nat hack to lock onto the first ip/port the dialog used.

make an acl called ast in acl.conf.xml with the asterisk box ip in it

add apply_nat_acl=ast to the sip profile params.

Then when anything from that acl matches FS will use the nat hacks to lock on the address
it sends packets to.



On Tue, Nov 4, 2008 at 11:55 AM, David Aldworth <daldworth@teliax.com (daldworth@teliax.com)> wrote:

[daldworth at teliax.com]

Anthony, In hopes of matching all IP's we added a very simple:

<list name="nat" default="allow">
</list>


To the acl.conf.xml and we added:


<param name="apply_nat_acl" value="nat"/>


To the sip profile. Unfortunately there was no affect. What would be the correct acl to match all IP's?


David


On Nov 4, 2008, at 1:18 PM, Anthony Minessale wrote:

[brian at freeswitch.org]

0.0.0.0/0 should match all IP space.

/b

On Nov 5, 2008, at 5:16 PM, David Aldworth wrote:

[daldworth at teliax.com]

Brian, we updated the acl to:

<list name="nat" default="allow">
<node type="allow" cidr="0.0.0.0/0"/>
</list>

And the ACK is still going to the wrong (right but wrong) ip/port.

Is there any way to get that ACK to go to the ip/port of the UDP header?

David

On Nov 5, 2008, at 4:21 PM, Brian West wrote:

[mike at jerris.com]

Please open a bug on http://jira.freeswitch.org . Please include a
full debug output from the freeswitch console with TPORT_LOG enabled
(info on the sofia page on the wiki).

Mike

On Nov 5, 2008, at 11:39 PM, David Aldworth wrote:

[anthony.minessale at g...]

did you remember to add
<param name="apply_nat_acl" value="nat"/>
to the profile in question and restart?

On Wed, Nov 5, 2008 at 10:39 PM, David Aldworth <daldworth@teliax.com (daldworth@teliax.com)> wrote:

[daldworth at teliax.com]

Yes. Below are settings that have been persistent through recent testing. Is there anything else we can try or should we open a jira?

<settings>
<param name="debug" value="0"/>
<param name="sip-trace" value="no"/>
<param name="rfc2833-pt" value="101"/>
<param name="sip-port" value="5060"/>
<param name="dialplan" value="XML"/>
<param name="context" value="public"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs" value="$${outbound_codec_prefs}"/>
<param name="hold-music" value="$${hold_music}"/>
<param name="use-rtp-timer" value="true"/>
<param name="rtp-timer-name" value="soft"/>
<param name="multiple-registrations" value="true"/>
<param name="manage-presence" value="true"/>
<param name="aggressive-nat-detection" value="true"/>
<param name="NDLB-force-rport" value="true"/>
<param name="inbound-codec-negotiation" value="generous"/>
<param name="nonce-ttl" value="60"/>
<param name="auth-calls" value="true"/>
<param name="rtp-timeout-sec" value="1800"/>
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="sip-ip" value="$${local_ip_v4}"/>
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
<param name="ext-sip-ip" value="$${external_sip_ip}"/>
<param name="rtp-timeout-sec" value="300"/>
<param name="rtp-hold-timeout-sec" value="1800"/>
<param name="apply_nat_acl" value="nat"/>
</settings>


On Nov 6, 2008, at 7:01 AM, Anthony Minessale wrote:

[anthony.minessale at g...]

doh,
I keep doing that sorry.

apply-nat-acl not apply_nat_acl



On Thu, Nov 6, 2008 at 8:22 AM, David Aldworth <daldworth@teliax.com (daldworth@teliax.com)> wrote: