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
©ó 2013/4/24 ¤U¤È 07:08, pratham2013 ´£¨ì:
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.
ThanksPhone : 919099923079
=====================
Manish Khilwani
Skype: manish_khilwani
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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