Try as much as you like, but it is not going to work :)
Asterisk does not include any attribute of the incoming INVITE to the
outgoing INVITE, so you will loose the profile-id and the h264 will be
only established in CIF. Enable the sip logs and verify it.
Best regards
Sergio
El 13/04/2011 1:19, Joel Wiramu Pauling escribió:
Hrm, setting either directmedia=yes and directrtpsetup=yes in
sip.conf does not seem to fix the issue.
I wonder if this is a network issue, everything is on routable
address's endpoint wise, and the gateway in between routes between my
RFC1918 address network (which the asterisk server sits on), i've done
this sort of setup in prod before tho and it works well.
On 13 April 2011 10:39, Joel Wiramu Pauling<j...@aenertia.net> wrote:
Cheers will give that a go, thanks for your input Gunnar.
wrt @amit: Codec is supported, it's the SDP/ATV combination ( I assume
that's the resolution ) that it is saying is unsupported - h264 ( the
codec for video ) is working fine, I think you are seeing the Siren
(audio) mismatches thats fine it falls back to ulaw.
Kind regards
-JoelW
On 13 April 2011 08:35, Gunnar Schaller<li...@nowin.de> wrote:
Hello,
Try a Dial without "tr" parameters and with "directmedia=yes" in
sip.conf.
http://www.voip-info.org/wiki/view/Asterisk+SIP+media+path
Regards,
Gunnar
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