[Freeswitch-users] INVITE ext-rtp-ip not set in FROM and SDP

Adam Kelloway adam.kelloway at newpace.ca
Wed Apr 25 20:01:53 MSD 2012


So this INVITE originates from your Lync Srv? When FreeSWITCH is used as 
a server, I believe it would detect that it doesn't need to use 
ext-rtp-ip because the INVITE originated inside the network. I have 
never used FreeSWITCH as a proxy, however. Have you checked out the 
wiki? http://wiki.freeswitch.org/wiki/Special:Search?search=SBC&go=Go

On 25/04/2012 6:46 AM, Martin Soerensen wrote:
>
> 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
>

-- 
Adam
-- 
NewPace Logo 	
	
	
	Adam Kelloway

	Software Engineer, NewPace
phone 	+1 (902) 406--8375  x1031
email 	Adam.Kelloway at NewPace.com <mailto:Adam.Kelloway at newpace.com>
aim <aim:GoIm?screenname=Adam.Kelloway at newpace.com>/msn 
<msnim:chat?contact=Adam.Kelloway at newpace.com> 	Adam.Kelloway 
<aim:GoIm?screenname=Adam.Kelloway at newpace.ca>@NewPace.ca

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