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@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@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@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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