[asterisk-bugs] [Asterisk 0018797]: RTP Early Media not Passed to Caller
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Feb 14 02:30:04 CST 2011
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=18797
======================================================================
Reported By: imp
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 18797
Category: Core/RTP
Reproducibility: always
Severity: minor
Priority: normal
Status: feedback
Asterisk Version: Older 1.6.2 - please test a newer version
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2011-02-11 10:44 CST
Last Modified: 2011-02-14 02:30 CST
======================================================================
Summary: RTP Early Media not Passed to Caller
Description:
In Switzerland the price of value added numbers is announced via early
audio before the connection is established to allow the caller to hang up
without generating costs.
This early media is not passed if the call is routed via asterisk.
Traces show:
Asterisk sends invite with sdp to carrier SIP GW.
SIP GW starts sending RTP early media to RTP endpoint specified by
invite.
SIP GW signals '100 trying'
SIP GW signals '180 ringing'
No Audio is passed to the client connected to the asterisk.
Depending on the 'progressinband' and 'rematuremedia' settings, either the
client is told to 'ring' or inband 'ringing' is generated by asterisk, but
no media from the carrier forwarded.
SIP GW signals '200 OK' + sdp
Call is established and two way audio is working. The caller has missed
the early media announcement.
According my interpretation of RFC3960 asterisk should forward early audio
if it is receiving early audio.
======================================================================
----------------------------------------------------------------------
(0131916) imp (reporter) - 2011-02-14 02:30
https://issues.asterisk.org/view.php?id=18797#c131916
----------------------------------------------------------------------
How trying to copy paste all sip transactions belongig to that call from
this very very busy asterisk:
<--- SIP read from UDP:157.161.3.19:5060 --->
INVITE sip:0901456056 at 157.161.10.35;user=phone SIP/2.0
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, SUBSCRIBE, OPTIONS,
UPDATE
Supported: replaces,timer,100rel
User-Agent: OmniPCX Enterprise R9.0 h1.301.40
Session-Expires: 1800;refresher=uac
Min-SE: 900
Content-Type: application/sdp
To: <sip:0901456056 at 157.161.10.35;user=phone>
From:
sip:0618269314 at vl-xm001001.imp.ch;tag=5a20ba5a28f8e172062f83d711ed55cf
Contact: <sip:157.161.3.19;transport=UDP>
all-ID: 5a734204cc6891c34586bd62243f8421 at 157.161.3.19
CSeq: 1349603881 INVITE
Via: SIP/2.0/UDP
157.161.3.19;branch=z9hG4bKf8249c380b529fa322eadb5692305976
Max-Forwards: 70
Content-Length: 393
v=0
o=OXE 1297671866 1297671866 IN IP4 157.161.3.19
s=abs
c=IN IP4 157.161.4.223
t=0 0
m=audio 32514 RTP/AVP 8 0 4 18 101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:30
a=rtpmap:0 PCMU/8000
a=ptime:20
a=maxptime:30
a=rtpmap:4 G723/8000
a=ptime:30
a=maxptime:30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=maxptime:40
a=rtpmap:101 telephone-event/8000
<------------->
--- (15 headers 21 lines) ---
Sending to 157.161.3.19 : 5060 (no NAT)
Using INVITE request as basis request -
5a734204cc6891c34586bd62243f8421 at 157.161.3.19
Found peer 'alcatel' for '0618269314' from 157.161.3.19:5060
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10d
(g723|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined -
0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 157.161.4.223:32514
Looking for 0901456056 in from-alcatel (domain 157.161.10.35)
list_route: hop: <sip:157.161.3.19;transport=UDP>
<--- Transmitting (no NAT) to 157.161.3.19:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
157.161.3.19;branch=z9hG4bKf8249c380b529fa322eadb5692305976;received=157.161.3.19
From:
sip:0618269314 at vl-xm001001.imp.ch;tag=5a20ba5a28f8e172062f83d711ed55cf
To: <sip:0901456056 at 157.161.10.35;user=phone>
Call-ID: 5a734204cc6891c34586bd62243f8421 at 157.161.3.19
CSeq: 1349603881 INVITE
Server: Asterisk PBX 1.6.2.15
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:0901456056 at 157.161.10.35>
Content-Length: 0
<------------>
-- Executing [0901456056 at from-alcatel:1] AGI("SIP/alcatel-0000aad1",
"screening-customer.agi,alcatel.conf") in new stack
-- Launched AGI Script
/usr/local/share/asterisk/agi-bin/screening-customer.agi
-- <SIP/alcatel-0000aad1>AGI Script screening-customer.agi completed,
returning 0
-- Executing [0901456056 at from-alcatel:2] AGI("SIP/alcatel-0000aad1",
"dial-imp-new-emerg.agi,4133,") in new stack
-- Launched AGI Script
/usr/local/share/asterisk/agi-bin/dial-imp-new-emerg.agi
-- AGI Script Executing Application: (Progress) Options: ()
Audio is at 157.161.10.35 port 14216
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (no NAT) to 157.161.3.19:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
157.161.3.19;branch=z9hG4bKf8249c380b529fa322eadb5692305976;received=157.161.3.19
From:
sip:0618269314 at vl-xm001001.imp.ch;tag=5a20ba5a28f8e172062f83d711ed55cf
To: <sip:0901456056 at 157.161.10.35;user=phone>;tag=as1017af28
Call-ID: 5a734204cc6891c34586bd62243f8421 at 157.161.3.19
CSeq: 1349603881 INVITE
Server: Asterisk PBX 1.6.2.15
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:0901456056 at 157.161.10.35>
Content-Type: application/sdp
Content-Length: 288
v=0
o=root 199946624 199946624 IN IP4 157.161.10.35
s=Asterisk PBX 1.6.2.15
c=IN IP4 157.161.10.35
t=0 0
m=audio 14216 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
-- AGI Script Executing Application: (SIPAddHeader) Options: (Privacy:
none)
-- <SIP/alcatel-0000aad1>AGI Script dial-imp-new-emerg.agi completed,
returning 0
-- Executing [0901456056 at from-alcatel:3] Macro("SIP/alcatel-0000aad1",
"dundi,41901456056") in new stack
-- Executing [s at macro-dundi:1] NoOp("SIP/alcatel-0000aad1", "DUNDi
Lookup 41901456056") in new stack
-- Executing [s at macro-dundi:2] Goto("SIP/alcatel-0000aad1",
"41901456056,1") in new stack
-- Goto (macro-dundi,41901456056,1)
-- Executing [0901456056 at from-alcatel:4] Dial("SIP/alcatel-0000aad1",
"SIP/41901456056 at imp1") in new stack
Audio is at 157.161.10.35 port 15708
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 157.161.10.10:5060:
INVITE sip:41901456056 at 157.161.10.10 SIP/2.0
Via: SIP/2.0/UDP 157.161.10.35:5060;branch=z9hG4bK130fc3a3;rport
Max-Forwards: 70
From: "41618269314" <sip:41618269314 at 157.161.10.35>;tag=as520dcb3b
To: <sip:41901456056 at 157.161.10.10>
Contact: <sip:41618269314 at 157.161.10.35>
Call-ID: 1c4ad0e32b89deb832dce3dd4bc50813 at 157.161.10.35
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.15
Date: Mon, 14 Feb 2011 08:24:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Privacy: none
Content-Type: application/sdp
Content-Length: 266
v=0
o=root 1730262329 1730262329 IN IP4 157.161.10.35
s=Asterisk PBX 1.6.2.15
c=IN IP4 157.161.10.35
t=0 0
m=audio 15708 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called 41901456056 at imp1
<--- SIP read from UDP:157.161.10.10:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
157.161.10.35:5060;branch=z9hG4bK130fc3a3;received=157.161.10.35;rport=5060
From: "41618269314" <sip:41618269314 at 157.161.10.35>;tag=as520dcb3b
To: <sip:41901456056 at 157.161.10.10>
Call-ID: 1c4ad0e32b89deb832dce3dd4bc50813 at 157.161.10.35
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:157.161.10.10:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
157.161.10.35:5060;branch=z9hG4bK130fc3a3;received=157.161.10.35;rport=5060
From: "41618269314" <sip:41618269314 at 157.161.10.35>;tag=as520dcb3b
To: <sip:41901456056 at 157.161.10.10>;tag=f1f26313
Call-ID: 1c4ad0e32b89deb832dce3dd4bc50813 at 157.161.10.35
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
-- SIP/imp1-0000aad2 is ringing
<--- SIP read from UDP:157.161.3.19:5060 --->
CANCEL sip:0901456056 at 157.161.10.35;user=phone SIP/2.0
Supported: replaces,timer,100rel
User-Agent: OmniPCX Enterprise R9.0 h1.301.40
Call-ID: 5a734204cc6891c34586bd62243f8421 at 157.161.3.19
To: <sip:0901456056 at 157.161.10.35;user=phone>
CSeq: 1349603881 CANCEL
From:
sip:0618269314 at vl-xm001001.imp.ch;tag=5a20ba5a28f8e172062f83d711ed55cf
Via: SIP/2.0/UDP
157.161.3.19;branch=z9hG4bKf8249c380b529fa322eadb5692305976
Max-Forwards: 70
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Sending to 157.161.3.19 : 5060 (no NAT)
<--- Reliably Transmitting (no NAT) to 157.161.3.19:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP
157.161.3.19;branch=z9hG4bKf8249c380b529fa322eadb5692305976;received=157.161.3.19
From:
sip:0618269314 at vl-xm001001.imp.ch;tag=5a20ba5a28f8e172062f83d711ed55cf
To: <sip:0901456056 at 157.161.10.35;user=phone>;tag=as1017af28
Call-ID: 5a734204cc6891c34586bd62243f8421 at 157.161.3.19
CSeq: 1349603881 INVITE
Server: Asterisk PBX 1.6.2.15
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 157.161.3.19:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
157.161.3.19;branch=z9hG4bKf8249c380b529fa322eadb5692305976;received=157.161.3.19
From:
sip:0618269314 at vl-xm001001.imp.ch;tag=5a20ba5a28f8e172062f83d711ed55cf
To: <sip:0901456056 at 157.161.10.35;user=phone>;tag=as1017af28
Call-ID: 5a734204cc6891c34586bd62243f8421 at 157.161.3.19
CSeq: 1349603881 CANCEL
Server: Asterisk PBX 1.6.2.15
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'1c4ad0e32b89deb832dce3dd4bc50813 at 157.161.10.35' in 32000 ms (Method:
INVITE)
Reliably Transmitting (no NAT) to 157.161.10.10:5060:
CANCEL sip:41901456056 at 157.161.10.10 SIP/2.0
Via: SIP/2.0/UDP 157.161.10.35:5060;branch=z9hG4bK130fc3a3;rport
Max-Forwards: 70
From: "41618269314" <sip:41618269314 at 157.161.10.35>;tag=as520dcb3b
To: <sip:41901456056 at 157.161.10.10>
Call-ID: 1c4ad0e32b89deb832dce3dd4bc50813 at 157.161.10.35
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.2.15
Content-Length: 0
---
Scheduling destruction of SIP dialog
'1c4ad0e32b89deb832dce3dd4bc50813 at 157.161.10.35' in 32000 ms (Method:
INVITE)
== Spawn extension (from-alcatel, 0901456056, 4) exited non-zero on
'SIP/alcatel-0000aad1'
<--- SIP read from UDP:157.161.10.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 157.161.10.35:5060;branch=z9hG4bK130fc3a3;rport
From: "41618269314" <sip:41618269314 at 157.161.10.35>;tag=as520dcb3b
To: <sip:41901456056 at 157.161.10.10>
Call-ID: 1c4ad0e32b89deb832dce3dd4bc50813 at 157.161.10.35
CSeq: 102 CANCEL
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:157.161.10.10:5060 --->
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP
157.161.10.35:5060;branch=z9hG4bK130fc3a3;received=157.161.10.35;rport=5060
From: "41618269314" <sip:41618269314 at 157.161.10.35>;tag=as520dcb3b
To: <sip:41901456056 at 157.161.10.10>;tag=f1f26313
Call-ID: 1c4ad0e32b89deb832dce3dd4bc50813 at 157.161.10.35
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Transmitting (no NAT) to 157.161.10.10:5060:
ACK sip:41901456056 at 157.161.10.10 SIP/2.0
Via: SIP/2.0/UDP 157.161.10.35:5060;branch=z9hG4bK130fc3a3;rport
Max-Forwards: 70
From: "41618269314" <sip:41618269314 at 157.161.10.35>;tag=as520dcb3b
To: <sip:41901456056 at 157.161.10.10>;tag=f1f26313
Contact: <sip:41618269314 at 157.161.10.35>
Call-ID: 1c4ad0e32b89deb832dce3dd4bc50813 at 157.161.10.35
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.15
Content-Length: 0
---
Really destroying SIP dialog
'1c4ad0e32b89deb832dce3dd4bc50813 at 157.161.10.35' Method: INVITE
<--- SIP read from UDP:157.161.3.19:5060 --->
ACK sip:0901456056 at 157.161.10.35;user=phone SIP/2.0
Call-ID: 5a734204cc6891c34586bd62243f8421 at 157.161.3.19
From:
sip:0618269314 at vl-xm001001.imp.ch;tag=5a20ba5a28f8e172062f83d711ed55cf
To: <sip:0901456056 at 157.161.10.35;user=phone>;tag=as1017af28
Via: SIP/2.0/UDP
157.161.3.19;branch=z9hG4bKf8249c380b529fa322eadb5692305976
CSeq: 1349603881 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Issue History
Date Modified Username Field Change
======================================================================
2011-02-14 02:30 imp Note Added: 0131916
======================================================================
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