[Spce-user] Call forwarding, no RTP

[Daniel Grotti]

Hi Alessandro,
it's very hard it's an ngcp issue, but to be sure please make an SIP/RTP
trace and check if NGCP is actually receiving the RTP stream from both
endpoints.


*Daniel Grotti *
VoIP Engineer

Sipwise GmbH <http://www.sipwise.com> , Campus 21/Europaring F15
AT-2345 Brunn am Gebirge

Phone:  +43(0)1 301 2032 <callto:+4313012032>
Email:  dgrotti at sipwise.com <mailto:dgrotti at sipwise.com>
Website:  www.sipwise.com <http://www.sipwise.com>

Particulars according Austrian Companies Code paragraph 14
"Sipwise GmbH" - Europaring F15 - 2345 Brunn am Gebirge
FN:305595f, Commercial Court Vienna, ATU64002206

On 01/19/2016 01:24 PM, stefanormc wrote:
> if this is still the issue:
>
>>             if I configure a subscriber to unconditionally forward
>>             any incoming call to an external number, get the
>>             forwarded call estabilished but I get no RTP flows on
>>             both sides.
>>
>
> then yes, we are talking about the same issue.
>
> try pressing any dtmf when the call is established on callee side. in
> my case rtp starts going through
>
>
>
>
>
>> On 19 Jan 2016, at 13:00, Alessandro Bolletta
>> <alessandro at mediaspot.net <mailto:alessandro at mediaspot.net>> wrote:
>>
>> No, i'm not using Digitel. Also I don't think that we are talking
>> about the same issue.
>>
>>
>> *Da:* stefanormc <stefanormc at gmail.com <mailto:stefanormc at gmail.com>>
>> *Inviato:* 19/gen/2016 12:36
>> *A:* Marco Teixeira
>> *Cc:* Alessandro Bolletta; spce-user at lists.sipwise.com
>> <mailto:spce-user at lists.sipwise.com>
>> *Oggetto:* Re: [Spce-user] Call forwarding, no RTP
>>
>> I’m experiencing the same problem. I have 2 providers: one works fine
>> the other not. Sipwise setting are the same.
>>
>> The non working provider is Digitel; is that the same of yours,
>> Alessandro?
>>
>> Apparently is a RFC compliance issue
>>
>> stefano
>>
>>> On 19 Jan 2016, at 12:34, Marco Teixeira <admin at marcoteixeira.com
>>> <mailto:admin at marcoteixeira.com>> wrote:
>>>
>>> So i think my initial assumption "(...) If you are allowing direct
>>> RTP between endpoints (...)" does not verify then.
>>> My next step would be to take a SIP trace and check what's going on...
>>>
>>>
>>> ---
>>> ​B​
>>> est regards
>>> Marco
>>> ---
>>>
>>>
>>> On Tue, Jan 19, 2016 at 11:24 AM, Alessandro Bolletta
>>> <alessandro at mediaspot.net <mailto:alessandro at mediaspot.net>> wrote:
>>>
>>>     That setting was already enabled.
>>>
>>>      
>>>
>>>     Alessandro
>>>
>>>      
>>>
>>>     *Da:*Marco Teixeira [mailto:admin at marcoteixeira.com
>>>     <mailto:admin at marcoteixeira.com>]
>>>     *Inviato:* martedì 19 gennaio 2016 12.12
>>>
>>>
>>>     *A:* Alessandro Bolletta <alessandro at mediaspot.net
>>>     <mailto:alessandro at mediaspot.net>>
>>>     *Cc:* spce-user at lists.sipwise.com
>>>     <mailto:spce-user at lists.sipwise.com>
>>>     *Oggetto:* Re: [Spce-user] Call forwarding, no RTP
>>>
>>>      
>>>
>>>     Goto your peering server for that "Ext number" 
>>>
>>>     Click Preferences
>>>
>>>     and set use_rtpproxy to always
>>>
>>>     I think that should do make the RTP for calls to that peer to
>>>     always proxy via SPCE
>>>
>>>     <image001.png>
>>>
>>>
>>>      
>>>
>>>     ---
>>>
>>>     Best regards
>>>
>>>     Marco
>>>
>>>     ---
>>>
>>>      
>>>
>>>      
>>>
>>>     On Tue, Jan 19, 2016 at 10:37 AM, Alessandro Bolletta
>>>     <alessandro at mediaspot.net <mailto:alessandro at mediaspot.net>> wrote:
>>>
>>>         Yes, the “Ext number” gateway refuses RTP from that endpoint
>>>         for sure.
>>>
>>>          
>>>
>>>         Could Sipwise be set in order to proxy that RTP flow?
>>>
>>>          
>>>
>>>         *Da:*Marco Teixeira [mailto:admin at marcoteixeira.com
>>>         <mailto:admin at marcoteixeira.com>]
>>>         *Inviato:* martedì 19 gennaio 2016 10.51
>>>         *A:* Alessandro Bolletta <alessandro at mediaspot.net
>>>         <mailto:alessandro at mediaspot.net>>
>>>         *Cc:* spce-user at lists.sipwise.com
>>>         <mailto:spce-user at lists.sipwise.com>
>>>         *Oggetto:* Re: [Spce-user] Call forwarding, no RTP
>>>
>>>          
>>>
>>>         If you are allowing direct RTP between endpoints, your
>>>         "external number" gateway/provider might be refusing RTP
>>>         from the original endpoint and expecting audio to come from
>>>         your SPCE.
>>>
>>>
>>>          
>>>
>>>         ---
>>>
>>>         Best regards
>>>
>>>         Marco
>>>
>>>         ---
>>>
>>>          
>>>
>>>          
>>>
>>>         On Mon, Jan 18, 2016 at 4:09 PM, Alessandro Bolletta
>>>         <alessandro at mediaspot.net <mailto:alessandro at mediaspot.net>>
>>>         wrote:
>>>
>>>             Hi,
>>>
>>>              
>>>
>>>             if I configure a subscriber to unconditionally forward
>>>             any incoming call to an external number, get the
>>>             forwarded call estabilished but I get no RTP flows on
>>>             both sides.
>>>
>>>              
>>>
>>>             The peer used for the call forwarding is already working
>>>             flawlessly for other purposes, so I can imagine that
>>>             there’s something with call forwarding’s procedure.
>>>
>>>              
>>>
>>>             Hoping to get your help,
>>>
>>>             Alessandro Bolletta
>>>
>>>
>>>             _______________________________________________
>>>             Spce-user mailing list
>>>             Spce-user at lists.sipwise.com
>>>             <mailto:Spce-user at lists.sipwise.com>
>>>             https://lists.sipwise.com/listinfo/spce-user
>>>
>>>          
>>>
>>>      
>>>
>>>
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>>
>
>
>
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