30 Apr
2013
30 Apr
'13
11:04 p.m.
JsSIP seems to do this now. Although it only creates a single flow.
Flow-tokens get added to Path: headers for REGISTERs, but they do not get
added to the Record-Route: headers for INVITEs (because the INVITE
contains a GRUU Contact: without an ;ob parameter). This is correct under
RFC5626.
To get in-dialog routing to work correctly for this stack (when GRUU and
outbound are enabled on Kamailio) you need to always do a location
lookup() when the R-URI is a GRUU URI.
In route[WITHINDLG] I now do something like:
if (has_totag()) {
if (is_gruu()) {
route[LOCATION];
} else if (loose_route()) {
} else {
}
exit;
}
Juha Heinanen writes:
--
Peter Dunkley
Technical Director
Crocodile RCS Ltd
yes, if in the example that i gave in previous
message p1 (through which
the call was setup) is still alive, but only connection to between ua1
and p1 has died, then bye from ua2 would reach ua1 from p1 to p2 to ua1
if p1 does lookup on the gruu of ua1 and after that does the
t_load_contact/t_next_contacts/t_next_contact_flow thing. in that case,
there would be no change in the route set as long as no flow tokens are
included in the r-r uris.
forgot to mention that i have not been able to test this outbound/gruu
scenario, because i don't know any sip client that would implement both
outbound and gruu.
-- juha
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