[Freeswitch-users] One way audio to CME

[Peter]

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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