Hi,

I have set these params.
It seems that the profile of internal and external are all the same as before and still no audio in port 5080.
Here are the profiles.

freeswitch@internal> sofia status profile internal
=================================================================================================
Name                    internal
Domain Name             N/A
Auto-NAT                false
DBName                  sofia_reg_internal
Pres Hosts              10.0.110.33,10.0.110.33
Dialplan                XML
Context                 public
Challenge Realm         auto_from
RTP-IP                  10.0.110.33
Ext-RTP-IP              59.120.xx.xx
SIP-IP                  10.0.110.33
URL                     sip:mod_sofia@10.0.110.33:5060
BIND-URL                sip:mod_sofia@10.0.110.33:5060
HOLD-MUSIC              local_stream://moh
OUTBOUND-PROXY          N/A
CODECS IN               G729,H264
CODECS OUT              G729,H264
TEL-EVENT               101
DTMF-MODE               rfc2833
CNG                     13
SESSION-TO              0
MAX-DIALOG              0
NOMEDIA                 false
LATE-NEG                true
PROXY-MEDIA             false
ZRTP-PASSTHRU           true
AGGRESSIVENAT           false
STUN-ENABLED            true
STUN-AUTO-DISABLE       false
CALLS-IN                0
FAILED-CALLS-IN         0
CALLS-OUT               0
FAILED-CALLS-OUT        0
REGISTRATIONS           0


freeswitch@internal> 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                  10.0.110.33
Ext-RTP-IP              59.120.xx.xx
SIP-IP                  10.0.110.33
Ext-SIP-IP              59.120.xx.xx
URL                     sip:mod_sofia@59.120.xx.xx:5080
BIND-URL                sip:mod_sofia@59.120.xx.xx:5080;maddr=10.0.110.33
HOLD-MUSIC              local_stream://moh
OUTBOUND-PROXY          N/A
CODECS IN               G729,H264
CODECS OUT              G729,H264
TEL-EVENT               101
DTMF-MODE               rfc2833
CNG                     13
SESSION-TO              0
MAX-DIALOG              0
NOMEDIA                 false
LATE-NEG                true
PROXY-MEDIA             false
ZRTP-PASSTHRU           true
AGGRESSIVENAT           false
STUN-ENABLED            true
STUN-AUTO-DISABLE       false
CALLS-IN                0
FAILED-CALLS-IN         0
CALLS-OUT               0
FAILED-CALLS-OUT        0
REGISTRATIONS           0


Any advice?

Thanks.
Jimmy


於 2013/4/24 下午 07:08, pratham2013 提到:
Hello Jimmy,

Put the bind-server-ip to auto and set your public ip to ext-rtp-ip and ext-sip-ip in vars.xml file. Hope it will work.

Thanks
=====================
Manish Khilwani
Skype: manish_khilwani
Gmail: [hidden email]
MSN : [hidden email]
Phone : 919099923079


On Wed, Apr 24, 2013 at 4:29 PM, Jimmy Chang [via freeswitch-users] <[hidden email]> wrote:
Hi,

In the default configuration, the internal profile listen port 5060 and external profile listen port 5080.
And follow the setting in http://wiki.freeswitch.org/wiki/NAT_Traversal
vars.xml
<X-PRE-PROCESS cmd="set" data="bind_server_ip=59.120.xx.xx"/>
<X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/>
<X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/>

internal.xml
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>

external.xml
<param name="ext-sip-ip" value="$${external_sip_ip}"/>
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>


When user make a SIP call from public network using external profile(5080), he encountered no audio.
Here is the SDP of 200 OK from FS to user.
v=0
o=FreeSWITCH 1366781127 1366781128 IN IP4 10.0.110.33
s=FreeSWITCH
c=IN IP4 10.0.110.33
t=0 0
m=audio 10568 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
m=video 10530 RTP/AVP 116
a=rtpmap:116 H264/90000
a=fmtp:116 profile-level-id=428014


When user make a SIP call from public network using internal profile(5060), everything OK.
Here is the SDP of 200 OK from FS to user.
v=0
o=FreeSWITCH 1366780920 1366780921 IN IP4 59.120.xx.xx
s=FreeSWITCH
c=IN IP4 59.120.xx.xx
t=0 0
m=audio 10892 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
m=video 10178 RTP/AVP 116
a=rtpmap:116 H264/90000
a=fmtp:116 profile-level-id=428014


I can't figure out which parameters in sofia profile or in dialplan make port 5060 set the public IP in SDP and work fine.
Any advice?

Thanks in advance.
Jimmy


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