oej at edvina.net
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 Posted: Tue Jan 08, 2008 11:24 am Post subject: [asterisk-users] Early media support for Asterisk behind NAT |
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8 jan 2008 kl. 07.41 skrev Mayur:
| Quote: | Hi,
I have asterisk 1.4.16 behind a NAT-FW which is using a hosted
SIP trunk for PSTN calling. Asterisk is configured to support nat
with nat=yes in sip.conf. Now the hosted PSTN Gateway supports
symmetric RTP and early media using 183 Session Progress. So If I
call a PSTN number which has IVR message played before the call is
connected (via 183), those media RTP packets do not reach the
asterisk inside till asterisk sends out media packet to the PSTN
gateway. I have used rtpkeepalive option and set it to 1 sec. But it
seems that I drop rtp voice packets in the initial instructions
played by the IVR.
How do I make sure that asterisk sends RTP packets (null rtp) to the
PSTN gateway just after receiving the media details in 183 SDP to
open the firewall port from inside?
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That's a very interesting question. We are able to receive media as
soon as we send the INVITE, but I am unsure on when we actually start
sending media. Turn on RTP debugging in your asterisk to check. I
would assume that if you have rtpkeepalive, we should start sending as
soon as we get somewhere to send to, which in this case is when we get
the SDP in the 183. There might be issues with some packets being sent
at the same time as the gateway sends 183. With QoS priority for
media, these may arrive to the NAT before the 183 SIP reply, which
will be a problem in this NATted situation.
There's no way we can actually send anything before the 183, so there
will always be time between SDP exchange and first functional media
packet in NAT situations. I always consider this when playing prompts
and wait at least a second before important audio begins.
/O |
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