[asterisk-users] RTP stream not passing through router with port forwarding

[Nasir Javaid]

Hi,

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
router.

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.

regards,

Nasir Javaid
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100803/06e05a6e/attachment.htm