[asterisk-users] RTP traffic not being forwarded

[Vivek Shrivastava]

Hi Ryan,

Are the SIP and RTP ports are randomly selected or there are specific ports
for these? Unchecking
random port selection option on the device/softphone may help.

--Vivek


On 11/10/07, Ryan Newington <ryan at lithiumblue.com> wrote:
>
> Hi Luki,
>
> Thanks for your advice. I've checked the firewall and it is set to allow
> all incoming traffic. I changed the media port range as well with no
> success.
>
> Some calls work fine. This is the configuration that doesn't work. The RTP
> traffic passes along the chain fine, but the Asterisk server doesn't do
> anything with the packets it gets from the near-end SIP phone and the media
> gateway.
>
> SIP Phone <-> Media Gateway <-> Asterisk <-> SIP Phone
>
> An asterisk internal call will work fine. Eg;
>
> SIP Phone <-> Asterisk <-> SIP Phone
>
> Regards
>
> Ryan
>
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] On Behalf Of Luki
> Sent: Sunday, 11 November 2007 12:52 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] RTP traffic not being forwarded
>
> > When using 'rtp debug' on the asterisk console, it shows that it is
> > receiving traffic from one endpoint, but not the other. A wireshark
> trace
> > reveals it is actually receiving traffic from both ends.
>
> Sounds like a firewall issue. Wireshark shows what's "on the wire",
> i.e. before iptables. The packets are being dropped for whatever
> reason and never reach the asterisk process. Check your iptables and
> RTP port range, and perhaps try changing it.
>
> Luki
>
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