Bugs item #2814137, was opened at 2009-06-29 19:29
Message generated for change (Comment added) made by marcushunger
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Category: modules
Group: ver 1.5.x
Status: Open
Resolution: None
Priority: 5
Private: No
Submitted By: Marcus Hunger (marcushunger)
Assigned to: Nobody/Anonymous (nobody)
Summary: force_rtp_proxy bug
Initial Comment:
force_rtp_proxy seems to handle re-invite wrong, resulting in one-way-audio.
----------------------------------------------------------------------
>Comment By: Marcus Hunger (marcushunger)
Date: 2009-07-01 11:14
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 08:53
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 15:15
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 11:45
Message:
happens when i handle a reinvite with rtpproxy_offer("l")
----------------------------------------------------------------------
Comment By: Daniel-Constantin Mierla (miconda)
Date: 2009-06-29 20:19
Message:
When is this happening?
----------------------------------------------------------------------
You can respond by visiting:
https://sourceforge.net/tracker/?func=detail&atid=743020&aid=2814137&group_…
Looks like seas uses a tm function which is not present in sr's tm
module. t_cancel_uac in K's tm module was only used by seas.
seas_action.c: In function 'ac_cancel':
seas_action.c:495: error: 'struct tm_binds' has no member named
't_cancel_uac'
seas_action.c: In function 'ac_reply':
seas_action.c:725: warning: passing argument 3 of
'seas_f.tmb.t_reply_with_body' from incompatible pointer type
seas_action.c:725: warning: passing argument 4 of
'seas_f.tmb.t_reply_with_body' from incompatible pointer type
seas_action.c:725: warning: passing argument 5 of
'seas_f.tmb.t_reply_with_body' from incompatible pointer type
seas_action.c:725: warning: passing argument 6 of
'seas_f.tmb.t_reply_with_body' from incompatible pointer type
seas_action.c: In function 'forward_sl_request':
regards
Klaus
siptrace uses a TM callback which is not preent in sr's tm module
Further flag_idx2mask is missing.
regards
klaus
make[1]: Entering directory
`/home/darilion/software/siprouter/sip-router/modules_k/siptrace'
gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align
-minline-all-stringops -falign-loops -ftree-vectorize -mtune=athlon64
-Wall -DNAME='"ser"' -DVERSION='"2.99.0-dev01-serk"' -DARCH='"i386"'
-DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.3.2"'
-D__CPU_i386 -D__OS_linux -DSER_VER=2099000
-DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP
-DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE
-DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST
-DUSE_NAPTR -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK
-DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM
-DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD
-DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H
-DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DHAVE_EPOLL -DHAVE_SIGIO_RT
-DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT
-DOPENSER_MOD_INTERFACE -DMOD_NAME='"siptrace"' -c siptrace.c -o siptrace.o
siptrace.c: In function 'mod_init':
siptrace.c:228: warning: implicit declaration of function 'flag_idx2mask'
siptrace.c: In function 'sip_trace':
siptrace.c:519: warning: passing argument 1 of 'search_next_avp' from
incompatible pointer type
siptrace.c:529: warning: passing argument 1 of 'search_next_avp' from
incompatible pointer type
siptrace.c: In function 'trace_onreq_in':
siptrace.c:578: error: 'TMCB_REQUEST_BUILT' undeclared (first use in
this function)
siptrace.c:578: error: (Each undeclared identifier is reported only once
siptrace.c:578: error: for each function it appears in.)
siptrace.c: In function 'trace_onreq_out':
regards
klaus
Module: sip-router
Branch: daniel/xavp
Commit: ef79b7beb75f3f4ed3a5fa1ba6c76363dc254e3e
URL: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=ef79b7b…
Author: Daniel-Constantin Mierla <miconda(a)gmail.com>
Committer: Daniel-Constantin Mierla <miconda(a)gmail.com>
Date: Tue Jun 30 12:26:56 2009 +0200
core: introducing xavp (eXtended AVP)
- design:
- one master list stored in core
- internal integer id to optimise the search in list (hash value over the string name)
- name stored as str and null terminated (can use regexp over it)
- string values stored as str and null terminated
- other types of value that can be used internally (not in config)
- long
- long long
- time
- xavp (or list of xavp)
- generic data structure
- for more see: http://sip-router.org/wiki/devel/xavp
- NOTE: this is not a replacement for existing AVPs, but an add-on, a new pseudo-variable class
---
xavp.c | 508 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
xavp.h | 95 ++++++++++++
2 files changed, 603 insertions(+), 0 deletions(-)
Diff: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commitdiff;h=ef7…
Bugs item #2814137, was opened at 2009-06-29 19:29
Message generated for change (Comment added) made by klaus_darilion
You can respond by visiting:
https://sourceforge.net/tracker/?func=detail&atid=743020&aid=2814137&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: Open
Resolution: None
Priority: 5
Private: No
Submitted By: Marcus Hunger (marcushunger)
Assigned to: Nobody/Anonymous (nobody)
Summary: force_rtp_proxy bug
Initial Comment:
force_rtp_proxy seems to handle re-invite wrong, resulting in one-way-audio.
----------------------------------------------------------------------
>Comment By: Klaus Darilion (klaus_darilion)
Date: 2009-07-01 08:53
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 15:15
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 11:45
Message:
happens when i handle a reinvite with rtpproxy_offer("l")
----------------------------------------------------------------------
Comment By: Daniel-Constantin Mierla (miconda)
Date: 2009-06-29 20:19
Message:
When is this happening?
----------------------------------------------------------------------
You can respond by visiting:
https://sourceforge.net/tracker/?func=detail&atid=743020&aid=2814137&group_…