15 Jan
2015
15 Jan
'15
1:38 p.m.
Dear all,
I have an on-reply route that needs to change the SDP version for the
reply coming in. The use case is that I have a mobile originated call
and there is some Ericsson switch that doesn't like it when the SDP
version is updated (in this case by asterisk) although nothing has
changed to the actual SDP (183 session progress and then OK.) Funny
thing is that Asterisk will actually drop a call if it receives a
re-INVITE with same version... That's why they invented
ignoresdpversion, but now it's the other way around.... :)
Mobile phone -> Ericsson MSC -> ACME packet -> (18X.4X.XXX.XX) Kamailio
(10.41.0.21) -> Asterisk
The issue is that the asterisk sends a reply 200 OK, with an updated
version because it already sent SDP for 183 session progress. This can
be patched in asterisk, but in my scenario I can unfortunately not do
that. Thus trying to fix this on Kamailio.
I am able to 'fix' this currently by performing a subst on the sdp owner
variable:
onreply_route[WITHSDP] {
if (has_body("application/sdp")) {
if(ds_is_from_list()) {
rtpproxy_answer("wrei");
*if(subst("/^o=someowner ([0-9]+) ([0-9]+) IN IP4
(.*)$/o=someowner \1 \1 IN IP4 \3/")) {*
xlog("L_INFO", "Fixed Asterisk incorrect
version number in SDP");
}
# tried the answer here as well, but that
corrupts it even more
}
else {
rtpproxy_answer("wrie");
}
}
exit;
}
However this corrupts the SDP:
v=0
o=tismi 652858233 652858233 IN IP4 10.41.0.21*18X.4X.XXX.XX*
s=Some server
c=IN IP4 18X.4X.XXX.XX
t=0 0
m=audio 57644 RTP/AVP 8 0 18 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=nortpproxy:yes
When I do not substitute the SDP looks perfectly fine and the external
address shows as the IN IP4. But of course the version is incremented:
v=0
o=tismi 1606876535 *1606876536* IN IP4 *18X.4X.XXX.XX*
s=Some server
c=IN IP4 18X.4X.XXX.XX
t=0 0
m=audio 55410 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=sdpmangled:yes
The ericsson is now accepting this (although it's corrupt, I know....
probably the ACME doing something funky with it), but it causes issues
with another unknown piece of equipment that fails on parsing the
session owner. I hope there is something wrong with my subst, but I'm
afraid I can not do this from the on_reply route because SDP is only
updated once it finishes?
I know it's dangerous to alter the session version like this, so I made
sure the Asterisk will never send a re-INVITE. Now I need a way to not
corrupt the o=
Kind regards,
Matthias van der Vlies