[Freeswitch-users] Wrong port on response

Fri Sep 19 19:27:37 EDT 2008

On Sep 19, 2008, at 6:08 PM, David Aldworth wrote:

> Hello -
>
> Got an issue with Freeswitch not responding on the port that the  
> initial request was made on. I'm not beyond believing that it is a  
> NAT or router issue except that I can register a Cisco phone from  
> another location or a softphone from the same location without any  
> problem. This Aastra just won't work for some reason.
>
> We have connectile-dysfuntion turned on. Otherwise we are using the  
> default profile settings. Auth is on (as you can see from the  
> below). Basically, the Reg request comes from port 41450, but  
> freeswitch responds on port 5060. Again, other UA's work fine, just  
> one Cisco and one Aastra from this site do not. Meanwhile a soft  
> phone from this site, and the same model cisco from another site do  
> not work.
>
> SIP dump and external profile are below. Thank you for any help. David
>

This isn't a bug.  If you notice the phone explicitly said in its  
contact for us to contact them via 192.168.1.192:5060 so you'll need  
to enable stun on the phone or rport.  If you were to enable rport on  
the aastra it would just work correctly.  If Aastra doesn't support  
RFC3581 or STUN then they are worthless phones just like Polycom.  Its  
not the registrars problem to fix your nat issues.  The phone should  
support RFC3581 (rport) or STUN and it would just work like the Snom's  
do.

Try adding this param to your sofia profile.  It will break cisco  
phones or any other phone that follows the sip spec.  This explicitly  
breaks RFC to accommodate broken phones.

<param name="NDLB-force-rport" value="true"/> in your sofia profile.

/b

> U 2008/09/19 16:51:45.145303 63.211.239.34:41450 -> 70.42.223.23:5060
> REGISTER sip:atl.teliax.net SIP/2.0.
> Via: SIP/2.0/UDP 192.168.1.192:5060;branch=z9hG4bKc1362dfab.
> Max-Forwards: 70.
> Content-Length: 0.
> To: Aastra Test <sip:pleasehelp(aastra)@atl.teliax.net>.
> From: Aastra Test  
> <sip:pleasehelp(aastra)@atl.teliax.net>;tag=cbf2963ddfab0d6.
> Call-ID: ecae0e043ee65c5c5340749a9d28235f at 192.168.1.192.
> CSeq: 288881481 REGISTER.
> Contact: Aastra Test  
> <sip:pleasehelp(aastra)@192.168.1.192:5060;transport=udp>;expires=300.
> Allow-Events: talk,hold,conference.
> Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO.
> Expires: 300.
> User-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/ 
> v3.2.8.45.
> .

> U 2008/09/19 16:51:45.145493 70.42.223.23:5060 -> 63.211.239.34:5060
> SIP/2.0 401 Unauthorized.
> Via: SIP/2.0/UDP  
> 192.168.1.192:5060;branch=z9hG4bKc1362dfab;received=63.211.239.34.
> From: Aastra Test  
> <sip:pleasehelp(aastra)@atl.teliax.net>;tag=cbf2963ddfab0d6.
> To: Aastra Test  
> <sip:pleasehelp(aastra)@atl.teliax.net>;tag=Sarvm1DjmU3Zg.
> Call-ID: ecae0e043ee65c5c5340749a9d28235f at 192.168.1.192.
> CSeq: 288881481 REGISTER.
> User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,  
> NOTIFY, REFER, UPDATE, REGISTER, INFO.
> Supported: 100rel, timer, precondition, path, replaces.
> WWW-Authenticate: Digest realm="atl.teliax.net",  
> nonce="d3538e86-9d86-dd11-82bf-001143e64915", algorithm=MD5,  
> qop="auth".
> Content-Length: 0.
>
>
> <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
> <profile name="external">
>   <!-- This profile is only for outbound registrations to providers  
> -->
>   <gateways>
>     <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
>   </gateways>
>
>   <domains>
>     <domain name="$${domain}" parse="true"/>
>   </domains>
>
>   <settings>
>     <param name="debug" value="0"/>
>     <param name="sip-trace" value="no"/>
>     <param name="rfc2833-pt" value="101"/>
>     <param name="sip-port" value="5060"/>
>     <param name="dialplan" value="XML"/>
>     <param name="context" value="public"/>
>     <param name="dtmf-duration" value="100"/>
>     <param name="codec-prefs" value="$${outbound_codec_prefs}"/>
>     <param name="hold-music" value="$${hold_music}"/>
>     <param name="use-rtp-timer" value="true"/>
>     <param name="rtp-timer-name" value="soft"/>
>     <param name="manage-presence" value="false"/>
>     <param name="aggressive-nat-detection" value="true"/>
>     <param name="inbound-codec-negotiation" value="generous"/>
>     <param name="nonce-ttl" value="60"/>
>     <param name="auth-calls" value="true"/>
>     <param name="rtp-timeout-sec" value="1800"/>
>     <param name="rtp-ip" value="$${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>
> </profile>
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