[Freeswitch-users] One way audio to CME

[Anthony McGarry]

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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> 
> 
> 
> 
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> 
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> 
> _________________________________________________________________________
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> 
> 
> 
> 
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