asterisk users - May 2017 - Asterisk 14.3.1 > 14.4.1 upgrade pjsip nat broken?

Hi first post, so hope I'm not violating protocol. 

Been using Asterisk as home phone and hobby use for nearly 10 years. I
love this project. 

Anyway, would someone mind verifying my pjsip.conf ?  This seems to work
well for 14.3.1 but I get no rtp into my natted Linphone when I upgrade
to 14.4.1. Other than that the phone registers properly on 14.4.1. 

I can provide a pjsip log as well, but for now I'll start with this. 

Asterisk is behind a shorewall firewall on a private natted network. It
has a single interface eth0. 

Relevant pjsip.conf: 

[transport-tls-nat]
type=transport
protocol=tls
method=sslv23 ;sslv23 enables tls1.2 because reasons
cert_file=XXX ;removed
priv_key_file=XXX ;removed
bind=0.0.0.0:5061
external_media_address=x.x.x.x ;public ip
external_signaling_address=x.x.x.x ;public ip
local_net=192.168.0.0/16

[endpoint-common](!)
type=endpoint
context=users
disallow=all
allow=g722,ulaw,h264
dtmf_mode=info

[endpoint-sdes](!)
media_encryption=sdes

[aor-common](!)
type=aor
remove_existing=yes
max_contacts=1
maximum_expiration=160
qualify_frequency=60 

[207](endpoint-common,endpoint-sdes)
;Linphone
callerid=Chris <PSTN number>
auth=207
aors=207
mailboxes=201 at default
use_avpf=yes
rtp_symmetric=yes
media_use_received_transport=yes
force_rport=yes

[207]
type=auth
auth_type=userpass
password=supersecretpassword
username=207

[207](aor-common)
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On Tue, May 30, 2017, at 01:07 PM, Christopher van de Sande
wrote:
> Hi first post, so hope I'm not violating protocol. 
> 
> Been using Asterisk as home phone and hobby use for nearly 10 years. I
> love this project. 
> 
> Anyway, would someone mind verifying my pjsip.conf ?  This seems to work
> well for 14.3.1 but I get no rtp into my natted Linphone when I upgrade
> to 14.4.1. Other than that the phone registers properly on 14.4.1. 
> 
> I can provide a pjsip log as well, but for now I'll start with this. 
> 
> Asterisk is behind a shorewall firewall on a private natted network. It
> has a single interface eth0. 
The actual SIP traffic (pjsip set logger on) would also be useful to see
exactly what is being exchanged and where it is going.

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

On Tue, May 30, 2017, at 01:55 PM, Christopher van de Sande
wrote:
> Cool, I've attached 2 sip trace examples, 14.3.1 and 14.4.1. Both are
> using identical configurations. 
> 
> The biggest difference I can see happens right after the UPDATE message
> from MicroSIP. 
> 
> The SDP headers on 14.4.1 issue the private ip instead of the public ip:
Yeah, something is indeed up. Please file an issue[1] with the logs and
configuration and network details.

[1] https://issues.asterisk.org/jira

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

Will do, thanks for the confirmation! 

Chris.

On 2017-05-30 18:08, Joshua Colp wrote:
> On Tue, May 30, 2017, at 01:55 PM, Christopher van de Sande wrote: 
> 
>> Cool, I've attached 2 sip trace examples, 14.3.1 and 14.4.1. Both
are
>> using identical configurations. 
>> 
>> The biggest difference I can see happens right after the UPDATE message
>> from MicroSIP. 
>> 
>> The SDP headers on 14.4.1 issue the private ip instead of the public
ip:
> 
> Yeah, something is indeed up. Please file an issue[1 [1]] with the logs and
> configuration and network details.
> 
> [1] https://issues.asterisk.org/jira
 

Links:
------
[1] https://issues.asterisk.org/jira
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