Bugs item #2814137, was opened at 2009-06-29 20:29
Message generated for change (Comment added) made by miconda
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Category: modules
Group: ver 1.5.x
>Status: Closed
>Resolution: Accepted
Priority: 5
Private: No
Submitted By: Marcus Hunger (marcushunger)
>Assigned to: Daniel-Constantin Mierla (miconda)
Summary: force_rtp_proxy bug
Initial Comment:
force_rtp_proxy seems to handle re-invite wrong, resulting in one-way-audio.
----------------------------------------------------------------------
>Comment By: Daniel-Constantin Mierla (miconda)
Date: 2009-07-02 12:33
Message:
Patch applied to 1.5. Thanks.
----------------------------------------------------------------------
Comment By: Marcus Hunger (marcushunger)
Date: 2009-07-01 12:14
Message:
sdp is in invite and 200 ok. the thing is, nearly the same config still
worked in 1.2-branch, but after an upgrade to 1.4 it stopped.
it happens always, no matter if the callee or caller does the reinvite.
the following trace shows what happens. the rtpproxy-port in invite and
reply is 41324. they must differ as they are used for different
media-streams.
217.10.1.1 - sip-proxy & rtpproxy
172.20.21.2 - sip-proxy doing rtpproxy_* stuff
217.10.66.165 - uac
217.10.1.2 - uas
U 217.10.1.1:5060 -> 172.20.21.2:5060
INVITE sip:3993942p0@217.10.1.2 SIP/2.0.
Via: SIP/2.0/UDP 217.10.1.1:5060;branch=z9hG4bK78fccbad7757B75A.
Via: SIP/2.0/UDP
192.168.234.24;rport=61020;received=217.10.66.165;branch=z9hG4bK78fccbad7757B75A.
From: "Erika" <sip:3993942e1@sipgate.de>;tag=D6DBB0FD-C198C6EA.
To: <sip:10@sipgate.de;user=phone>;tag=as3804b1ae.
Route: <sip:172.20.21.2;lr=on>.
CSeq: 4 INVITE.
Call-ID: e521bbee-f129e0af-f6e95884(a)192.168.234.24.
Contact: <sip:3993942e1@217.10.66.165:61020>.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER.
User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.1.2.0078.
Supported: 100rel,replaces.
Allow-Events: talk,hold,conference.
Max-Forwards: 69.
Content-Type: application/sdp.
Content-Length: 297.
X-hint: rr-enforced.
X-nathint: nat.
.
v=0.
o=- 1246290867 1246290869 IN IP4 192.168.234.24.
s=Polycom IP Phone.
c=IN IP4 192.168.234.24.
t=0 0.
m=audio 2262 RTP/AVP 9 0 8 18 101.
a=sendrecv.
a=rtpmap:9 G722/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=direction:active.
#
U 172.20.21.2:5060 -> 217.10.1.1:5060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP 217.10.1.1:5060;branch=z9hG4bK78fccbad7757B75A.
Via: SIP/2.0/UDP
192.168.234.24;rport=61020;received=217.10.66.165;branch=z9hG4bK78fccbad7757B75A.
From: "Erika" <sip:3993942e1@sipgate.de>;tag=D6DBB0FD-C198C6EA.
To: <sip:10@sipgate.de;user=phone>;tag=as3804b1ae.
CSeq: 4 INVITE.
Call-ID: e521bbee-f129e0af-f6e95884(a)192.168.234.24.
Content-Length: 0.
.
#
U 172.20.21.2:5060 -> 217.10.1.2:5060
INVITE sip:3993942p0@217.10.1.2 SIP/2.0.
Via: SIP/2.0/UDP 172.20.21.2;branch=z9hG4bKe2fc.9dcfa89.0.
Via: SIP/2.0/UDP 217.10.1.1:5060;branch=z9hG4bK78fccbad7757B75A.
Via: SIP/2.0/UDP
192.168.234.24;rport=61020;received=217.10.66.165;branch=z9hG4bK78fccbad7757B75A.
From: "Erika" <sip:3993942e1@sipgate.de>;tag=D6DBB0FD-C198C6EA.
To: <sip:10@sipgate.de;user=phone>;tag=as3804b1ae.
CSeq: 4 INVITE.
Call-ID: e521bbee-f129e0af-f6e95884(a)192.168.234.24.
Contact: <sip:3993942e1@217.10.66.165:61020>.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER.
User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.1.2.0078.
Supported: 100rel,replaces.
Allow-Events: talk,hold,conference.
Max-Forwards: 68.
Content-Type: application/sdp.
Content-Length: 315.
X-hint: rr-enforced.
X-nathint: nat.
.
v=0.
o=- 1246290867 1246290869 IN IP4 192.168.234.24.
s=Polycom IP Phone.
c=IN IP4 217.10.1.1.
t=0 0.
m=audio 41324 RTP/AVP 9 0 8 18 101.
a=sendrecv.
a=rtpmap:9 G722/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=direction:active.
a=nortpproxy:yes.
#
U 217.10.1.2:5060 -> 172.20.21.2:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP
172.20.21.2;branch=z9hG4bKe2fc.9dcfa89.0;received=172.20.21.2.
Via: SIP/2.0/UDP 217.10.1.1:5060;branch=z9hG4bK78fccbad7757B75A.
Via: SIP/2.0/UDP
192.168.234.24;rport=61020;received=217.10.66.165;branch=z9hG4bK78fccbad7757B75A.
From: "Erika" <sip:3993942e1@sipgate.de>;tag=D6DBB0FD-C198C6EA.
To: <sip:10@sipgate.de;user=phone>;tag=as3804b1ae.
Call-ID: e521bbee-f129e0af-f6e95884(a)192.168.234.24.
CSeq: 4 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces, timer.
Contact: <sip:3993942p0@217.10.1.2>.
Content-Length: 0.
.
#
U 217.10.1.2:5060 -> 172.20.21.2:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
172.20.21.2;branch=z9hG4bKe2fc.9dcfa89.0;received=172.20.21.2.
Via: SIP/2.0/UDP 217.10.1.1:5060;branch=z9hG4bK78fccbad7757B75A.
Via: SIP/2.0/UDP
192.168.234.24;rport=61020;received=217.10.66.165;branch=z9hG4bK78fccbad7757B75A.
From: "Erika" <sip:3993942e1@sipgate.de>;tag=D6DBB0FD-C198C6EA.
To: <sip:10@sipgate.de;user=phone>;tag=as3804b1ae.
Call-ID: e521bbee-f129e0af-f6e95884(a)192.168.234.24.
CSeq: 4 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces, timer.
Contact: <sip:3993942p0@217.10.1.2>.
Content-Type: application/sdp.
Content-Length: 329.
.
v=0.
o=root 647471219 647471221 IN IP4 217.10.1.2.
s=sipgate VoIP GW.
c=IN IP4 217.10.1.2.
t=0 0.
m=audio 17732 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.
##
U 172.20.21.2:5060 -> 217.10.1.1:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 217.10.1.1:5060;branch=z9hG4bK78fccbad7757B75A.
Via: SIP/2.0/UDP
192.168.234.24;rport=61020;received=217.10.66.165;branch=z9hG4bK78fccbad7757B75A.
From: "Erika" <sip:3993942e1@sipgate.de>;tag=D6DBB0FD-C198C6EA.
To: <sip:10@sipgate.de;user=phone>;tag=as3804b1ae.
Call-ID: e521bbee-f129e0af-f6e95884(a)192.168.234.24.
CSeq: 4 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces, timer.
Contact: <sip:3993942p0@217.10.1.2>.
Content-Type: application/sdp.
Content-Length: 347.
.
v=0.
o=root 647471219 647471221 IN IP4 217.10.1.2.
s=sipgate VoIP GW.
c=IN IP4 217.10.1.1.
t=0 0.
m=audio 41324 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=nortpproxy:yes.
----------------------------------------------------------------------
Comment By: Klaus Darilion (klaus_darilion)
Date: 2009-07-01 09:53
Message:
do you have a SIP trace? Is it maybe related to late offer (SDP in 200 OK
and ACK)? Does the bug happens always or only if the reINVITE is sent by
the callee?
----------------------------------------------------------------------
Comment By: Nobody/Anonymous (nobody)
Date: 2009-06-30 16:15
Message:
in detail, the sdp is rewritten with the wrong port for this direction of
the session. it's the same as the answer's which does not work. to build
the query for the rtp-proxy from- and to-tag the have to be reversed in
this case.
----------------------------------------------------------------------
Comment By: Marcus Hunger (marcushunger)
Date: 2009-06-30 12:45
Message:
happens when i handle a reinvite with rtpproxy_offer("l")
----------------------------------------------------------------------
Comment By: Daniel-Constantin Mierla (miconda)
Date: 2009-06-29 21:19
Message:
When is this happening?
----------------------------------------------------------------------
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Revision: 5891
http://openser.svn.sourceforge.net/openser/?rev=5891&view=rev
Author: miconda
Date: 2009-07-02 09:32:45 +0000 (Thu, 02 Jul 2009)
Log Message:
-----------
- fix #2814137 - handle re-invite with force_rtp_proxy("l");
- patch by Marcus Hunger
Modified Paths:
--------------
branches/1.5/modules/nathelper/nathelper.c
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Bugs item #2813924, was opened at 2009-06-29 14:55
Message generated for change (Comment added) made by miconda
You can respond by visiting:
https://sourceforge.net/tracker/?func=detail&atid=743020&aid=2813924&group_…
Please note that this message will contain a full copy of the comment thread,
including the initial issue submission, for this request,
not just the latest update.
Category: modules
Group: ver 1.5.x
>Status: Closed
>Resolution: Accepted
Priority: 5
Private: No
Submitted By: Marcus Hunger (marcushunger)
>Assigned to: Daniel-Constantin Mierla (miconda)
Summary: nathelper crash
Initial Comment:
publishing very long payload-type to the rtp-proxy overflows a buffer. see nathelper.c +2758.
v[1].iov_len must be smaller than sizeof(opts)
----------------------------------------------------------------------
>Comment By: Daniel-Constantin Mierla (miconda)
Date: 2009-07-02 12:01
Message:
Thanks! Committed to 1.5.
----------------------------------------------------------------------
Comment By: Marcus Hunger (marcushunger)
Date: 2009-06-30 12:41
Message:
ah, sorry, you're right.
----------------------------------------------------------------------
Comment By: Daniel-Constantin Mierla (miconda)
Date: 2009-06-30 12:06
Message:
Did you meant '&&' operator instead of ',' in for condition? Otherwise:
warning: left-hand operand of comma expression has no effect
----------------------------------------------------------------------
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Revision: 5890
http://openser.svn.sourceforge.net/openser/?rev=5890&view=rev
Author: miconda
Date: 2009-07-02 09:00:06 +0000 (Thu, 02 Jul 2009)
Log Message:
-----------
- fix #2813924: publishing very long payload-type to the rtp-proxy overflows a buffer
- patch by Marcus Hunger
Modified Paths:
--------------
branches/1.5/modules/nathelper/nathelper.c
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Hi!
in ser's osp module Makefile these condition should be adopted to search
in /usr/lib too:
LIBS=$(shell if [ -f /usr/local/lib/libosptk.a ]; then echo "-losptk" ;
else echo "-losp" ; fi)
Or make it like K's osp module: use always -losptk
btw: shouldn't be the module rather identical in ser and K? Maybe
someone from the transnexus can handle this?
regards
klaus
Hello,
in kamailio there is a difference between exit and drop, while in ser is
basically the same (exit functionality).
For kamailio, in request and failure route they are the same, stop
processing of the actions. But in branch route and reply route drop does
a bit more: drop current processed branch respectively reply, so they
are not forwarded.
Addition of this functionality will affect core and tm. Any comments
regarding this? i can create the patch for it, if nobody sees issues.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://www.asipto.com/
i changed
NAME=srcmd
in utils/sercmd/Makefile
but that alone didn't make it, because main Makefile has
# which utils need compilation (directory path) and which to install
# (full path including file name)
utils_compile= utils/gen_ha1 utils/sercmd
utils_bin_install= utils/gen_ha1/gen_ha1 utils/sercmd/sercmd
and make results in error:
make[2]: Entering directory `/usr/src/trunk-src/sip-router/utils/sercmd'
gcc -Wall -g -O2 -DNAME='"srcmd"' -DVERSION='"0.2"' -DUSE_READLINE -c parse_listen_id.c -o parse_listen_id.o
gcc -Wall -g -O2 -DNAME='"srcmd"' -DVERSION='"0.2"' -DUSE_READLINE -c sercmd.c -o sercmd.o
gcc -Wl,-O2 -Wl,-E parse_listen_id.o sercmd.o -lresolv -lsctp -lreadline -lncurses -o srcmd
make[2]: Leaving directory `/usr/src/trunk-src/sip-router/utils/sercmd'
ERROR: utils/sercmd/sercmd not compiled
what is the proper fix?
my suggestion is to add to Makefile.defs a new variable SHORT_NAME,
which could default to ser:
SHORT_NAME=ser
and then use that Makefile:
utils_bin_install= utils/gen_ha1/gen_ha1 utils/sercmd/$(SHORTNAME)cmd
and also in utils/sercmd/Makefile:
NAME=$(SHORT_NAME)cmd
and also in modules_s/ctl/ctl_defaults.h instead of ser:
#define DEFAULT_CTL_SOCKET "unixs:/tmp/ser_ctl"
-- juha