[asterisk-users] Can asterisk support using different ip for rtp?

Klaus Darilion klaus.mailinglists at pernau.at
Thu Jun 26 03:22:29 CDT 2008


I think this is not possible. If you take a look at main/rtp.c there is 
no config option for an IP address.

regards
klaus

Jun Yin schrieb:
> some vendors(like alcatel-lucent) developed a kind of sip proxy which
> includes two parts: one sip signaling module and one or more voice
> modules. voice modules are responsible for receiving/sending voice
> traffic(RTP). each voice module has its own IP. so , when the sip
> signaling part sends out "invite" packet, it has sip ip in its sip
> content and different RTP ip in SDP content. (also for 200OK)
> Now I'm trying to do a test to simulate that product with asterisk. I
> hope asterisk can sends out different rtp address based on user or
> domain name. Based on network side, there are many ways to do it: we
> can configure the network card with multiple IPs, one for SIP and
> others for RTP.  or , we can setup multiple network cards for the
> asterisk server, one card is for sip signaling and other cards for rtp
> traffic connecting to different carriers.   I think this diagram is
> reasonable but I was surprised that asterisk does not support it.
> Maybe asterisk can do this by special configuration? or, there is
> other free sip proxy software can do this?
> 
> Thanks.
> 
>> Message: 10
>> Date: Wed, 25 Jun 2008 05:15:29 -0400
>> From: "Raj Jain" <rj2807 at gmail.com>
>> Subject: Re: [asterisk-users] Can asterisk support using different ip
>>        for     rtp?
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>>        <asterisk-users at lists.digium.com>
>> Message-ID:
>>        <1971b0b60806250215r22c31131l3886a762bc31855f at mail.gmail.com>
>> Content-Type: text/plain; charset="iso-8859-1"
>>
>> On Tue, Jun 24, 2008 at 9:26 PM, Jun Yin <hansyin at gmail.com> wrote:
>>
>>> Currently, RTP IP have to be the same as SIP IP. But, SIP RFC allows
>>> RTP to use different IP as SIP ip.
>>>
>>> Is there any way to configure it? GUI or CLI? or , will we support it in
>>> future?
>>>
>> SIP is decoupled from RTP, so they can emanate from different IP addresses.
>> Can you present a scenario where this will make sense (in the context where
>> Asterisk is anchoring the media) ?
>>
>> --
>> Raj Jain
> 
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