[sr-dev] [ openser-Bugs-2814137 ] force_rtp_proxy bug

Wed Jul 1 11:14:13 CEST 2009

Bugs item #2814137, was opened at 2009-06-29 19:29
Message generated for change (Comment added) made by marcushunger
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https://sourceforge.net/tracker/?func=detail&atid=743020&aid=2814137&group_id=139143

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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 at 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 at sipgate.de>;tag=D6DBB0FD-C198C6EA.
To: <sip:10 at sipgate.de;user=phone>;tag=as3804b1ae.
Route: <sip:172.20.21.2;lr=on>.
CSeq: 4 INVITE.
Call-ID: e521bbee-f129e0af-f6e95884 at 192.168.234.24.
Contact: <sip:3993942e1 at 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 at sipgate.de>;tag=D6DBB0FD-C198C6EA.
To: <sip:10 at sipgate.de;user=phone>;tag=as3804b1ae.
CSeq: 4 INVITE.
Call-ID: e521bbee-f129e0af-f6e95884 at 192.168.234.24.
Content-Length: 0.
.

#
U 172.20.21.2:5060 -> 217.10.1.2:5060
INVITE sip:3993942p0 at 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 at sipgate.de>;tag=D6DBB0FD-C198C6EA.
To: <sip:10 at sipgate.de;user=phone>;tag=as3804b1ae.
CSeq: 4 INVITE.
Call-ID: e521bbee-f129e0af-f6e95884 at 192.168.234.24.
Contact: <sip:3993942e1 at 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 at sipgate.de>;tag=D6DBB0FD-C198C6EA.
To: <sip:10 at sipgate.de;user=phone>;tag=as3804b1ae.
Call-ID: e521bbee-f129e0af-f6e95884 at 192.168.234.24.
CSeq: 4 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces, timer.
Contact: <sip:3993942p0 at 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 at sipgate.de>;tag=D6DBB0FD-C198C6EA.
To: <sip:10 at sipgate.de;user=phone>;tag=as3804b1ae.
Call-ID: e521bbee-f129e0af-f6e95884 at 192.168.234.24.
CSeq: 4 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces, timer.
Contact: <sip:3993942p0 at 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 at sipgate.de>;tag=D6DBB0FD-C198C6EA.
To: <sip:10 at sipgate.de;user=phone>;tag=as3804b1ae.
Call-ID: e521bbee-f129e0af-f6e95884 at 192.168.234.24.
CSeq: 4 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces, timer.
Contact: <sip:3993942p0 at 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.

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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?

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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.

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Comment By: Marcus Hunger (marcushunger)
Date: 2009-06-30 11:45

Message:
happens when i handle a reinvite with rtpproxy_offer("l")

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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_id=139143