[Freeswitch-users] IP infomation in SDP
Manish Khilwani
manish.khilwani at gmail.com
Wed Apr 24 15:52:03 MSD 2013
Hello Jimmy,
Can you let me now below things first.
1) Are you able to make calls with one extension to another extension?
2) if yes then please send me extensions and dialplan
*Thanks
=====================
Manish Khilwani
Skype: manish_khilwani
Gmail: manish.khilwani at gmail.com
MSN : manish.khilwani at hotmail.com
Phone : 919099923079*
On Wed, Apr 24, 2013 at 5:09 PM, Jimmy Chang <chang33.tw at gmail.com> wrote:
> 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 at 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 at 10.0.110.33:5060
> BIND-URL sip:mod_sofia at 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 at 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 at 59.120.xx.xx:5080
> BIND-URL sip:mod_sofia at 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]<http://user/SendEmail.jtp?type=node&node=7590072&i=0>
> MSN : [hidden email]<http://user/SendEmail.jtp?type=node&node=7590072&i=1>
> Phone : 919099923079*
>
>
> On Wed, Apr 24, 2013 at 4:29 PM, Jimmy Chang [via freeswitch-users] <[hidden
> email] <http://user/SendEmail.jtp?type=node&node=7590072&i=2>> 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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