[Freeswitch-users] One way audio to CME
Peter
eidevm5 at gmail.com
Thu Aug 8 06:29:46 MSD 2013
Hi Brian.
I did have the IP set in the external profile, but after removing them it
didn't seem to make any difference, ie: the SIP ports were listening on
the correct IP address.
Have to say I'm getting a little confused as to when and where to use
sip-ip and ext-sip-ip
Should sip-ip ALWAYS be set to your internal IP address and ext-sip-ip to
your external IP address?
If so, which profiles do they need to be set in? If they are set in both
the internal and external SIP profiles, does that matter?
Thanks
Peter
On Thu, Aug 8, 2013 at 11:41 AM, Brian Foster <bdfoster at davri.com> wrote:
> Your ip for the external profile will show whatever the ip is set to in
> the external profile configs. It doesn't matter what is set in the internal
> profile.
>
> Thank you,
>
> Brian Foster
> Project Manager/Owner's Rep.
> Davri Investments, Inc.
> O: 317-787-2686 x2102
> M: 317-600-9753
> E: bdfoster at davri.com
> Indianapolis, Indiana
>
> Sent from a mobile device.
> On Aug 7, 2013 9:36 PM, "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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>>
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>> http://www.freeswitchsolutions.com
>>
>>
>>
>>
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