i’d like to have 2 xlite stations communicate w/ each other directly w/o packets passing through the
asterisk. (of course, after sip does it’s thing.)
i set canreinvite=yes, nat=no, dtmfmode=sip info.
works fine w/ codecs supported by both xlite and asterisk.
doesn’t work w/ codecs not supported on asterisk.
in the later case, i get ‘failed to extablish call’.
since the codec comes into play only for rtp, why do need a codec that is supported also on the
asterisk?
with the setting described above, rtp packets are going directly between the 2 xlite stations.
Becaue Asterisk is a back to back user agent. In particular, it creates an internal bit map of the supported codecs to pass between the incoming and outgoing legs, and it needs to know the position in that bitmap to set for each different codec.
thanks for the reply david.
do you know of any way to config asterisk which frees me from dependency on the codecs
which it supports?
your explaination about the bitmap is very simple, but this was a coding or protocol decision.
ah, it suggests that if i can somehow allocate space in the bitmap for some codec which
it doesn’t really support, but is supported by the 2 endpoint stations, it could click.
(i realize i’m short on details here and there’s alot packed into that last statement.)
we’re trying to have students compare various codecs under stress conditions.
one that we want in that comparison is opus, which is supported by xlite, but not asterisk.
Adding new codecs is something you should discuss on the developer mailing list.
Some other things to note are that: canreinvite is deprecated in that version. Please read UPGRADE.txt for the replacement.
Also, at least some versions of X-Lite drop re-invite requests on the floor, rather than, at least, rejecting them. X-Lite is not a good choice for anything involving re-invites.
Finally, on my version of X-Lite, I see no option to use SIP INFO. I suspect it only supports RFC 2833, which means you can’t re-invite if you enable features that need DTMF detection.
i’m using xlite 4.5.5. don’t see any sip info param their either
btw, had the same problem w/ a pair of snom300 ip phones.
couldn’t even establish a session for a codec supported on the snom’s, but not the asterisk.
for purposes of the lab experiment, when we want to demo the different codecs, we don’t need
any pbx features. we just want to demo the codecs 1 to the other.
do u know of any other free soft ip phone which i may succeed with.
i don’t want to (but may have to) just start trying various products at random…
yes, i understand that in principle it should be doable.
the problem, as usual, is translating the principle to practice.
i keep u up on my progress (which i hope happens).
tnx
ams