[asterisk-users] RTP stream not passing through router with port forwarding
shmaltz at gmail.com
Tue Aug 3 06:21:06 CDT 2010
Is asterisk and the SIP device behind the same router?
Most routers will not redirect internal NAT requests. So that if you
are trying to have port forwarding done but the request and the
forwarding destination are on the same interface it won't work.
On 8/3/10, Nasir Javaid <nasirjavaidnasir at gmail.com> wrote:
> I am trying to dial a registered user via his IP:Port mechanism, but problem
> is that the audio data is not reaching to dialed user. here is the scenario.
> caller and callee both are registered at asterisk server. asterisk server is
> on public ip so no port forwarding and natting necessary there. however
> caller and callee both are behind router and there is port forwarding
> enabled and nat=yes, qualify=yes in sip.conf for both users.
> callee user name: adf
> callee local ip/port: 192.168.0.10:5678
> callee router ip: 116.79.x.x
> when we simply dial callee as Dial(SIP/adf) RTP stream reaches perfectly
> fine to 192.168.0.10 through router and INVITE is sent to local ip through
> INVITE sip:adf at 192.168.0.10:5678 SIP/2.0 (asterisk somehow manages to
> contact local ip through router and sends rtp there)
> but problem arises when i dial using IP:Port combination like this
> Dial(SIP/adf at 116.79.x.x:5678)
> In this case INVITE is sent to router ip instead of local ip through router.
> INVITE sip:adf at 116.79.x.x:5678 SIP/2.0 (asterisk sends rtp to router ip
> and not local ip)
> Similerly TO header also has same ip as INVITE. I think in IP dial rtp is
> not reaching to local ip through router as INVTE is meant for router ip and
> asterisk does not know where to send rtp stream after sending it to router.
> how can this issue be resolved? is there something to be done at router
> confiurations or sip.conf parameters. I have already played with
> nat/qualify/canreinvite/directrtp/externip etc parameters.
> Nasir Javaid
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