[asterisk-users] need help on setup rtp directly between 2 sip clients

[Alyed]

>so the asterisk is in middle in all version, right? thank you for your
explanation
is the one whom everyone goes and says "hey I'm 101 and live downstairs can
I play with you guys?"

>my goal is asterisk is on internet - WAN IP address and the softphones are
in NAT but the xlite supports the ICE function that is >why i ask the media
should be go directly between softphones and no need go through the asterisk


I guess you still don't fully get it :)

The scenario you mention is similar (at least for the "direct call" thingy)
ICE doesn't mean you don't need to know where the callee is it just means it
will play a little with the SDP part of the SIP. Have a look at
http://www.voiptraversal.com/ice_methodology.htm to better understand what's
ICE about.

Alyed


2010/3/26 haloha <haloha201 at gmail.com>

> Hi Alyed
>
> so the asterisk is in middle in all version, right? thank you for your
> explanation
> all devices i mean are asterisk + softphones
> my goal is asterisk is on internet - WAN IP address and the softphones are
> in NAT but the xlite supports the ICE function that is why i ask the media
> should be go directly between softphones and no need go through the asterisk
>
>
> will check the SJphone feature, thank you for your suggestion
>
>
> Thank you
>
> On Sat, Mar 27, 2010 at 9:04 AM, Alyed <alyed at vivoxie.com> wrote:
>
>> If your sofphones are registering to the asterisk, then asterisk needs to
>> be in the middle, otherwise there's no way your 101 sofpthone user can
>> actually know where (by where I mean which IP) is the 102 softphone user.
>>
>> UNLESS (yes, there's a big unless) you dial from 101 DIRECTLY to 102. How?
>> well dialing directly to 102's IP.That's where Xlite doesn't work, but
>> SJphone does.
>>
>> SJphone supports the advanced SIP URI syntax which for a user is:
>> sip:username at the.user.ip
>>
>> Nevertheless...... if you are inside a LAN, why wouldn't you want those
>> calls to go through asterisk??? If you have collision problems I suggest you
>> fix them instead of asking everyone to call using SIP uri.
>>
>> Alyed
>>
>>
>> 2010/3/26 haloha <haloha201 at gmail.com>
>>
>>> Hi Alyed
>>>
>>>
>>> xilte softphone work perfectly on other sip server(opensips server)
>>>
>>> Don't remember the exact syntax but guess it's something like
>>> sip:username at the.phones.ip:
>>>>
>>>> 5060
>>>
>>>
>>> >>>you mean i config the extension.conf look like exten =>
>>> 1000,1,Dial(SIP/1000 at ip address:5060), is it right?
>>>
>>> the problem i got here is the asterisk server to stay middle of  media
>>> first, then redirect the media later, how to fix it,asterisk no need stay in
>>> middle of media because all devices are in the same LAN
>>>
>>> is there another hint
>>>
>>> Thank you
>>>
>>>
>>> On Fri, Mar 26, 2010 at 11:56 PM, Alyed <alyed at vivoxie.com> wrote:
>>>
>>>> I guess to do what you want you need to dial directly between the
>>>> phones. Can't do it with xlite but you can with SJphones
>>>>
>>>> Don't remember the exact syntax but guess it's something like
>>>> sip:username at the.phones.ip:5060
>>>>
>>>> Alyed
>>>>
>>>>
>>>>
>>>
>>> --
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>>
>>
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>
>
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