[Freeswitch-users] One way audio to CME

Anthony McGarry agtmcgarry at gmail.com
Thu Aug 8 13:01:47 MSD 2013


Yes internal should show 10.10.10.206
change ext-sip-ip and ext-rtp-ip to "auto-nat" 
Restart and check sofia status

From the wiki
ext-rtp-ip
This is the IP behind which FreeSWITCH is seen from the Internet, so if FreeSWITCH is behind NAT, this is basically the public IP that should be used for RTP.

On 8 Aug 2013, at 02:30, Peter <eidevm5 at gmail.com> wrote:

> Hi Anthony
> 
> Really appreciate the taking your time to look at this.  It's starting to drive me nuts.
> 
> My dialplan for calls to CME is:
> 
>   <extension name="incoming-cme">
>         <condition field="destination_number" expression="^(3\d\d\d)$">
>             <action application="bridge" data="sofia/internal/${destination_number}@10.10.10.203"/>
>         </condition>
>   </extension>
> 
> 
> My internal sip profile has:
> 
>     <param name="rtp-ip" value="10.10.10.206"/>
>     <param name="sip-ip" value="10.10.10.206"/>
>     <param name="ext-rtp-ip" value="10.1.1.206"/>
>     <param name="ext-sip-ip" value="10.1.1.206"/>
> 
> The output from:
> 
> sofia status
> 
> is:
> 
> internal    profile                 sip:mod_sofia at 10.1.1.206:5060    RUNNING (0)
> external    profile                 sip:mod_sofia at 10.1.1.206:5060    RUNNING (0)
> 
> Should internal show 10.10.10.206??
> 
> The output from:
> 
> sofia status profile internal
> 
> shows:
> 
> Name                 internal
> Domain Name          N/A
> Auto-NAT             false
> DBName               sofia_reg_internal
> Pres Hosts           10.10.10.206,10.1.1.206
> Dialplan             XML
> Context              public
> Challenge Realm      auto_from
> RTP-IP               10.10.10.206
> Ext-RTP-IP           10.1.1.206
> SIP-IP               10.10.10.206
> Ext-SIP-IP           10.1.1.206
> URL                  sip:mod_sofia at 10.1.1.206:5060
> BIND-URL             sip:mod_sofia at 10.1.1.206:5060;maddr=10.10.10.206
> HOLD-MUSIC           N/A
> OUTBOUND-PROXY       N/A
> CODECS IN            iLBC at 30i,PCMU,PCMA,GSM
> CODECS OUT           iLBC at 30i,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
> ZRTP-PASSTHRU        false
> AGGRESSIVENAT        false
> STUN-ENABLED         true
> STUN-AUTO-DISABLE    false
> 
> 
> The SIP trace from the Freeswitch SBC is at:
> 
> http://pastebin.freeswitch.org/21279
> 
> I've been playing around with all sorts of different combinations of SIP/RTP IP settings, but still no closer.
> 
> Hope you have some insight.
> 
> Thanks
> 
> Peter
> 
> On Wed, Aug 7, 2013 at 6:31 PM, Anthony McGarry <agtmcgarry at gmail.com> wrote:
> Hi Peter,
> 
> Your debug shows the invite with via/from/contact/rpid all coming from 10.1.1.206, your external side. 
> Check your bridge statement, is it using the correct sip profile? 
> Check your sip profile SBC internal params rtp-ip & sip-ip, make sure they are set correctly to 10.10.10.206
> Paste up your logs from the sbc including sip trace.
> 
> Anthony
> 
> 
> On 7 Aug 2013, at 08:12, Peter <eidevm5 at gmail.com> wrote:
> 
>> Hi Anthony.
>> 
>> Yes, the SIP profiles are the same for calls going to Kamailio and to CME/CUBE.
>> 
>> Note that CME only has one interface, so binding the source interface doesn't really make much sense.
>> 
>> Note that I've simplified my set up a little and the phone that was registered to CUCM is now registered to CME.  However, the result is still the same, ie:  one way audio to the Cisco phone.
>> 
>> You can see the SIP debug from CME at:
>> 
>> http://pastebin.freeswitch.org/21274
>> 
>> The call is coming from 1001 at 10.1.1.204 to 3000 at 10.10.10.203
>> 
>> where
>> 
>> 10.1.1.204  - Freeswitch where SIP clients register to
>> 10.1.1.206  - External side of Freeswitch SBC
>> 10.10.10.206 - Internal side of Freeswitch SBC
>> 10.10.10.203 - CME
>> 
>> Peter
>> 
>> 
>> 
>> On Tue, Aug 6, 2013 at 5:13 PM, Anthony McGarry <agtmcgarry at gmail.com> wrote:
>> Hi Peter,
>> 
>> Because the calls are fine when using Kamailio I'm assuming your sip profiles are fine and you FS SBC config is fine. Are you using the same profiles?
>> Yes you are correct. Have you added the commands? Add them as a first step.
>> Send on a 'debug ccsip messages' 
>> 
>> Anthony 
>> 
>> 
>> 
>> On 6 Aug 2013, at 05:35, Peter <eidevm5 at gmail.com> wrote:
>> 
>>> Thanks for replying Anthony.
>>> 
>>> Keep in mind that I have very little experience with Cisco products, so I may be missing something fundamental.
>>> 
>>> As far as I can see
>>> 
>>> voice-class sip bind media source-interface ....
>>> 
>>> is just used to bind the SIP or media stream to the appropriate interface on the CUBE.
>>> 
>>> My issue is that the CUBE is trying to initiate the return RTP stream to the external interface (instead of the internal interface) on the Freeswitch SBC.
>>> 
>>> Is my understanding of the sip bind media command correct?
>>> 
>>> Thanks
>>> 
>>> Peter
>>> 
>>> 
>>> On Mon, Aug 5, 2013 at 5:23 PM, Anthony McGarry <agtmcgarry at gmail.com> wrote:
>>> On cube make sure you specify the source address on your dial-peers
>>> voice-class sip bind media|control
>>> to the correct side. I have seen one way audio when not set.
>>> 
>>> On 5 Aug 2013, at 06:29, Peter <eidevm5 at gmail.com> wrote:
>>> 
>>> >
>>> >
>>> > I currently have successful two way calls (signalling and media) in the following setup
>>> >
>>> >
>>> > External Linphone -->  Freeswitch  --> Freeswitch SBC  -> Router  -> Kamailio --> Internal Linphone
>>> >
>>> > However, when I try to call a Cisco handset that is registered to CUCM9 via CME in the following config:
>>> >
>>> > External Linphone -->  Freeswitch  --> Freeswitch SBC  -> Router  -> CME ->  CUCM9 -->  Cisco handset
>>> >
>>> > The call signalling appears to be working fine and I can successfully  initiate a call from each end, but the only RTP stream that is working is from the external Linphone client to the Cisco handset.
>>> >
>>> > Note that CME is being used as a CUBE device, so all SIP and RTP goes via it.
>>> >
>>> > Looking at the RTP debugs on CME I can see the problem is that the "Media Dest Addr" is getting set to the external side of the FS SBC rather than the internal IP address.
>>> >
>>> >
>>> > I tried setting adding:
>>> >
>>> >             <action application="set" data="disable_rtp_auto_adjust="true" />
>>> >
>>> > to the dialplan on the SBC, but it made no difference.
>>> >
>>> >
>>> > Any suggestions as to what to check next?
>>> >
>>> > Thanks
>>> >
>>> > Peter
>>> >
>>> > _________________________________________________________________________
>>> 
>>> 
>> 
>>> _________________________________________________________________________
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>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
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>> 
>> _________________________________________________________________________
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>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
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>> 
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>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
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>> 
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> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
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> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
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> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
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> 
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