[Freeswitch-users] One way audio to CME
Peter
eidevm5 at gmail.com
Thu Aug 8 05:30:13 MSD 2013
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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>>
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> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
>
>
>
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> http://wiki.freeswitch.org
> http://www.cluecon.com
>
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>
> _________________________________________________________________________
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>
>
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