[Freeswitch-users] INVITE ext-rtp-ip not set in FROM and SDP
Martin Soerensen
martin.soerensen at kontron.com
Wed Apr 25 13:46:42 MSD 2012
Hi all. Hope that somebody can help my...
First.......... I'm new to freeswitch... so please be patient ;o)
I'm trying to use freeswitch as a SBC.. lync isn't playing nice with NAT... so I'm sending all my call through FS trying to rewrite SIP headers
I'm having problems with ext-rtp-ip and SDP in my INVITE send by freeswitch.... the external IP is not set in FROM and SDP when I dial out via lync.. I can set FROM by setting <param name="from-domain" value="xxx.xxx.157.92"/> but SDP is still my internal IP address
My setup :
10.70.0.182 10.70.0.59 xxx.xxx.157.92 yyy.yyy.65.50
Lync Srv <------------------> FS <---------------> Firewall Nat <----------------> ISP/SIP gateway
FreeSWITCH Version 1.1.beta1 (git-2a25c4f 2012-04-24 16-30-20 +0200)
SIP trace:
send 1078 bytes to udp/[yyy.yyy.65.50]:5060 at 07:50:03.294218:
------------------------------------------------------------------------
INVITE sip:+4529802936 at yyy.yyy.65.50:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.157.92;rport;branch=z9hG4bK9FZjDgDrUyHvD
Max-Forwards: 69
From: "Martin Soerensen" <sip:+4578772660 at 10.70.0.59>;tag=7vjX1tBj987XS
To: <sip:+4529802936 at yyy.yyy.65.50:5060>
Call-ID: 1b96e79a-094e-1230-a29b-000c29236c5d
CSeq: 27334037 INVITE
Contact: <sip:gw+gc-gateway at xxx.xxx.157.92:5060;transport=udp;gw=gc-gateway>
User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-2a25c4f 2012-04-24 16-30-20 +0200
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 201
X-FS-Support: update_display,send_info
Remote-Party-ID: "Martin Soerensen" <sip:+4578772660 at 10.70.0.59>;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1335307501 1335307502 IN IP4 10.70.0.59
s=FreeSWITCH
c=IN IP4 10.70.0.59
t=0 0
m=audio 32702 RTP/AVP 0 8 3 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
------------------------------------------------------------------------
sofia status profile external
=================================================================================================
Name external
Domain Name N/A
Auto-NAT false
DBName sofia_reg_external
Pres Hosts
Dialplan XML
Context public
Challenge Realm auto_to
RTP-IP 10.70.0.59
Ext-RTP-IP xxx.xxx.157.92
SIP-IP 10.70.0.59
Ext-SIP-IP xxx.xxx.157.92
URL sip:mod_sofia at xxx.xxx.157.92:5060
BIND-URL sip:mod_sofia at xxx.xxx.157.92:5060;maddr=10.70.0.59
HOLD-MUSIC local_stream://moh
OUTBOUND-PROXY N/A
CODECS IN G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
CODECS OUT PCMU,PCMA,GSM
TEL-EVENT 101
DTMF-MODE rfc2833
CNG 13
SESSION-TO 0
MAX-DIALOG 0
NOMEDIA false
LATE-NEG false
PROXY-MEDIA false
AGGRESSIVENAT false
STUN-ENABLED true
STUN-AUTO-DISABLE false
CALLS-IN 0
FAILED-CALLS-IN 0
CALLS-OUT 2
FAILED-CALLS-OUT 2
REGISTRATIONS 0
sofia status gateway gc-gateway
=================================================================================================
Name gc-gateway
Profile external
Scheme Digest
Realm yyy.yyy.65.50:5060
Username User
Password yes
>From <sip:User at yyy.yyy.65.50:5060>
Contact <sip:gw+gc-gateway at 1xxx.xxx.157.92:5060;transport=udp;gw=gc-gateway>
Exten User
To sip:User at yyy.yyy.65.50:5060
Proxy sip:yyy.yyy.65.50:5060
Context public
Expires 3600
Freq 3600
Ping 1335339925
PingFreq 15
PingState -1/1/1
State NOREG
Status UP
CallsIN 0
CallsOUT 0
FailedCallsIN 0
FailedCallsOUT 0
=================================================================================================
External profle:
<profile name="external">
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
<!-- This profile is only for outbound registrations to providers -->
<gateways>
<X-PRE-PROCESS cmd="include" data="external/*.xml"/>
</gateways>
<domains>
<domain name="all" alias="false" parse="true"/>
</domains>
<settings>
<param name="debug" value="0"/>
<param name="sip-trace" value="no"/>
<param name="sip-capture" value="no"/>
<param name="rfc2833-pt" value="101"/>
<param name="sip-port" value="$${external_sip_port}"/>
<param name="dialplan" value="XML"/>
<param name="context" value="public"/>
<param name="dtmf-duration" value="2000"/>
<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
<param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
<param name="hold-music" value="$${hold_music}"/>
<param name="rtp-timer-name" value="soft"/>
<param name="local-network-acl" value="localnet.auto"/>
<param name="manage-presence" value="false"/>
<!--<param name="aggressive-nat-detection" value="true"/>-->
<param name="apply-nat-acl" value="rfc1918"/>
<param name="inbound-codec-negotiation" value="generous"/>
<param name="nonce-ttl" value="60"/>
<param name="auth-calls" value="false"/>
<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>
Gateway
<?xml version="1.0"?>
<include>
<gateway name="gc-gateway">
<param name="realm" value="yyy.yyy.65.50:5060"/>
<param name="username" value="User"/>
<param name="password" value="Password"/>
<param name="register" value="false"/>
<param name="caller-id-in-from" value="true"/>
<param name="ping" value="15"/>
</gateway>
</include>
Best regards
Martin
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120425/9f9b6e90/attachment-0001.html
Join us at ClueCon 2011 Aug 9-11, 2011
More information about the FreeSWITCH-users
mailing list