[asterisk-bugs] [Asterisk 0013958]: SDP replies incorrect - 'a=inactive' - replied to with 'a=sendrecv'
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Dec 10 02:14:50 CST 2008
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=13958
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Reported By: toc
Assigned To: mnicholson
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Project: Asterisk
Issue ID: 13958
Category: Channels/chan_sip/General
Reproducibility: always
Severity: minor
Priority: normal
Status: acknowledged
Asterisk Version: 1.6.0
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
======================================================================
Date Submitted: 2008-11-24 05:15 CST
Last Modified: 2008-12-10 02:14 CST
======================================================================
Summary: SDP replies incorrect - 'a=inactive' - replied to
with 'a=sendrecv'
Description:
When asterisk is sent a packet from Microsoft OCS Mediation Server, where
the call is being placed on hold, Asterisk sends an incorrect response.
Mediation server is 192.168.1.208
Asterisk is 192.168.1.250 (on 5061).
OpenSIPS is intercepting to resolve the bug on 192.168.1.250 (on 5060)
The packet sent from OCS is below (intercepted using OpenSIPS):
[ Method INVITE from 192.168.1.208:1276 (2) ]
INVITE sip:01234567 at 192.168.1.250:5061 SIP/2.0
FROM: <sip:First.Last at domain.com>;epid=A4325290A1;tag=f7768ee4ad
TO: <sip:01234567 at 192.168.1.250;user=phone>;tag=as79279f7b
CSEQ: 69 INVITE
CALL-ID: 1ae1e5ef-f96f-482c-9c44-da92c9094ecc
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 192.168.1.208:1276;branch=z9hG4bK1d297960
CONTACT:
<sip:domain.com.au:5060;transport=Tcp;maddr=192.168.1.208;ms-opaque=2cea147ecbefbb7f>
CONTENT-LENGTH: 268
SUPPORTED: 100rel
USER-AGENT: RTCC/3.0.0.0 MediationServer
CONTENT-TYPE: application/sdp; charset=utf-8
v=0
o=- 0 0 IN IP4 192.168.1.208
s=session
c=IN IP4 192.168.1.208
b=CT:1000
t=0 0
m=audio 60806 RTP/AVP 0 101
c=IN IP4 192.168.1.208
a=rtcp:60807
a=inactive
a=label:Audio
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
[ End of Request (2) ]
Response from Asterisk:
[ Reply 200 (OK) from 192.168.1.250:5061 concerning INVITE ]
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.250;branch=z9hG4bK3795.6ca29175.0;i=2;received=192.168.1.250
Via: SIP/2.0/TCP 192.168.1.208:1276;branch=z9hG4bK1d297960
From: <sip:First.Last at domain.com.au>;epid=A4325290A1;tag=f7768ee4ad
To: <sip:01234567 at 192.168.1.250;user=phone>;tag=as79279f7b
Call-ID: 1ae1e5ef-f96f-482c-9c44-da92c9094ecc
CSeq: 69 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:01234567 at 192.168.1.250:5061>
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 295744575 295744576 IN IP4 192.168.1.250
s=Asterisk PBX 1.6.0
c=IN IP4 192.168.1.250
t=0 0
m=audio 19848 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
[ End of Reply ]
The asterisk packet contains an incorrect a=sendrecv, it should be
a=inactive as OCS has requested the call to be placed on hold.
The hold works when modifying the packet from a=sendrecv to a=inactive
using OpenSIPS.
======================================================================
----------------------------------------------------------------------
(0096067) toc (reporter) - 2008-12-10 02:14
http://bugs.digium.com/view.php?id=13958#c96067
----------------------------------------------------------------------
Ok - reverted to the asterisk + OCS mediation server (no patch applied):
More feedback pending the patch being applied.
* SIP Call
1. Rx INVITE / 232 INVITE / sip:<external
number>@192.168.1.250;user=phone
2. NewChan Channel SIP/network.domain.com.au-08bc0b48 - fr
3. TxResp SIP/2.0 / 232 INVITE - 100 Trying
4. TxResp SIP/2.0 / 232 INVITE - 180 Ringing
5. TxRespRel SIP/2.0 / 232 INVITE - 200 OK
6. Rx ACK / 232 ACK / sip:<external
number>@192.168.1.250:5060;transport=TCP
7. Rx INVITE / 233 INVITE / sip:<external
number>@192.168.1.250:5060;transpo
8. ReInv Re-invite received
9. TxResp SIP/2.0 / 233 INVITE - 100 Trying
10. TxRespRel SIP/2.0 / 233 INVITE - 200 OK
11. Rx ACK / 233 ACK / sip:<external
number>@192.168.1.250:5060;transport=TCP
# pressed hold
test*CLI>
* SIP Call
1. Rx INVITE / 232 INVITE / sip:<external
number>@192.168.1.250;user=phone
2. NewChan Channel SIP/network.domain.com.au-08bc0b48 - fr
3. TxResp SIP/2.0 / 232 INVITE - 100 Trying
4. TxResp SIP/2.0 / 232 INVITE - 180 Ringing
5. TxRespRel SIP/2.0 / 232 INVITE - 200 OK
6. Rx ACK / 232 ACK / sip:<external
number>@192.168.1.250:5060;transport=TCP
7. Rx INVITE / 233 INVITE / sip:<external
number>@192.168.1.250:5060;transpo
8. ReInv Re-invite received
9. TxResp SIP/2.0 / 233 INVITE - 100 Trying
10. TxRespRel SIP/2.0 / 233 INVITE - 200 OK
11. Rx ACK / 233 ACK / sip:<external
number>@192.168.1.250:5060;transport=TCP
12. Rx INVITE / 234 INVITE / sip:<external
number>@192.168.1.250:5060;transpo
13. ReInv Re-invite received
14. TxResp SIP/2.0 / 234 INVITE - 100 Trying
15. TxRespRel SIP/2.0 / 234 INVITE - 200 OK
16. Rx ACK / 234 ACK / sip:<external
number>@192.168.1.250:5060;transport=TCP
And here's another, doing exact the same, with SIP debug (note a=inactive
from OCS, replied to with a=sendrecv by Asterisk):
<--- SIP read from TCP://192.168.1.208:1335 --->
INVITE sip:<external number>@192.168.1.250:5060;transport=TCP SIP/2.0
FROM:
<sip:First.Last at network.domain.com.au>;epid=A4325290A1;tag=16f16e814
TO: <sip:<external number>@192.168.1.250;user=phone>;tag=as1d515069
CSEQ: 236 INVITE
CALL-ID: 7ca84582-7c97-4706-a3da-70068a647fd3
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 192.168.1.208:1335;branch=z9hG4bKd2f9eb80
CONTACT:
<sip:Mediation.network.domain.com.au:5060;transport=Tcp;maddr=192.168.1.208;ms-opaque=2cea147ecbefbb7f>
CONTENT-LENGTH: 268
SUPPORTED: 100rel
USER-AGENT: RTCC/3.0.0.0 MediationServer
CONTENT-TYPE: application/sdp; charset=utf-8
v=0
o=- 0 0 IN IP4 192.168.1.208
s=session
c=IN IP4 192.168.1.208
b=CT:1000
t=0 0
m=audio 60472 RTP/AVP 0 101
c=IN IP4 192.168.1.208
a=rtcp:60473
a=inactive
a=label:Audio
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
0
a=rtpmap:8 PCMA/8000
a=ptime:20
<------------->
--- (12 headers 18 lines) ---
Sending to 192.168.1.208 : 1335 (no NAT)
test*CLI>
<--- Transmitting (no NAT) to 192.168.1.208:1335 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP
192.168.1.208:1335;branch=z9hG4bKd2f9eb80;received=192.168.1.208
From:
<sip:First.Last at network.domain.com.au>;epid=A4325290A1;tag=16f16e814
To: <sip:<external number>@192.168.1.250;user=phone>;tag=as1d515069
Call-ID: 7ca84582-7c97-4706-a3da-70068a647fd3
CSeq: 236 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:<external number>@192.168.1.250:5060;transport=TCP>
Content-Length: 0
<------------>
Audio is at 192.168.1.250 port 12568
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
test*CLI>
<--- Reliably Transmitting (no NAT) to 192.168.1.208:1335 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP
192.168.1.208:1335;branch=z9hG4bKd2f9eb80;received=192.168.1.208
From:
<sip:First.last at network.domain.com.au>;epid=A4325290A1;tag=16f16e814
To: <sip:<external number>@192.168.1.250;user=phone>;tag=as1d515069
Call-ID: 7ca84582-7c97-4706-a3da-70068a647fd3
CSeq: 236 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:<external number>@192.168.1.250:5060;transport=TCP>
Content-Type: application/sdp
Content-Length: 259
v=0
o=root 83368207 83368207 IN IP4 192.168.1.250
s=Asterisk PBX 1.6.0
c=IN IP4 192.168.1.250
t=0 0
m=audio 12568 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Issue History
Date Modified Username Field Change
======================================================================
2008-12-10 02:14 toc Note Added: 0096067
======================================================================
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