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Summary:ASTERISK-17393: RTP Early Media not Passed to Caller
Reporter:Benoit IMP Panizzon (imp)Labels:
Date Opened:2011-02-11 10:44:27.000-0600Date Closed:2011-07-26 14:28:43
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Core/RTP
Versions:1.6.2.15 Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) sip2.pcap.gz
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.

****** ADDITIONAL INFORMATION ******

http://www.faqs.org/rfc/rfc3960.txt

     1. Unless a 180 (Ringing) response is received, never generate
        local ringing.

     2. If a 180 (Ringing) has been received but there are no incoming
        media packets, generate local ringing.

     3. If a 180 (Ringing) has been received and there are incoming
        media packets, play them and do not generate local ringing.
Comments:By: Leif Madsen (lmadsen) 2011-02-11 12:22:51.000-0600

Not enough information provided to proceed.

Per the bug guidelines:

* SIP trace from the Asterisk console
* Enable debug level logging and SIP history via sip.conf
* Provide a configuration which can be used to duplicate the situation
* What is the topology in use and what piece of equipment

The configuration and the SIP trace are the most important.
By: Ronald Chan (loloski) 2011-02-11 15:03:23.000-0600

on my test rig/setup both SVN 1.8 i can't replicate this, please see attached snippet diaplan.

[from-peer]

exten => 11108123232,1,Progress
same => n,Ringing
same => n,Wait(2)
same => n,Playback(early-media,noanswer)
same => n,Dial(SIP/100,60,tT)
same => n,Voicemail(100@default)

of course i can not attest if the other end is not asterisk :D

just my 0.002 $

Regards
By: Benoit IMP Panizzon (imp) 2011-02-14 02:20:48.000-0600

sip.conf:

progressinband=no
prematuremedia=no

[alcatel]
type=peer
context=from-alcatel
host=157.161.3.19
qualify=yes
allow=alaw
language=de
dtmfmode=RFC2833
canreinvite=no

[imp1]
type=peer
context=from-imp
host=157.161.10.10
dtmfmode=auto
disallow=all
allow=alaw
language=de
canreinvite=no

----------------------
extensions.conf

[from-alcatel]
exten => _xx.,1,AGI(screening-customer.agi,alcatel.conf)
exten => _xx.,n,AGI(dial-imp-new-emerg.agi,4133,${SIP_HEADER(Privacy)})
exten => _xx.,n,Macro(dundi,${ITUNUM})
exten => _xx.,n,Dial(${IMPDESTINATION})
exten => _xx.,n,HangUp()

'screening-customer.agi' just makes sure the callerID the customer is signaling is realy belonging to the customer if not the caller id is replaced by the main number of the customer.

'dial-imp-new-emerg.agi' checks if the customer has dialed an emergency number which has to be routed to the nearest emergency central according to the postal code passed as argument. I'm ommiting that part.

From that AGI the important parts are:

$AGI->exec('Progress'); (I did try with and without, no success).

if ($privacy eq "user") {
       $AGI->exec('SIPAddHeader','Privacy: user; header; session');
       $AGI->exec('SIPAddHeader','P-Asserted-Identity: "ImproWare AG" <sip:+41618269300@157.161.3.19>');
} else {
       $AGI->exec('SIPAddHeader','Privacy: none');
}

$AGI->set_variable('IMPDESTINATION',"SIP/$exten\@imp1");
if ($exten =~ /^00(.*)/) {
       $exten = $1;
}
$AGI->set_variable('ITUNUM',"$exten");
By: Benoit IMP Panizzon (imp) 2011-02-14 02:30:03.000-0600

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@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@157.161.10.35;user=phone>
From: sip:0618269314@vl-xm001001.imp.ch;tag=5a20ba5a28f8e172062f83d711ed55cf
Contact: <sip:157.161.3.19;transport=UDP>
all-ID: 5a734204cc6891c34586bd62243f8421@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@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@vl-xm001001.imp.ch;tag=5a20ba5a28f8e172062f83d711ed55cf
To: <sip:0901456056@157.161.10.35;user=phone>
Call-ID: 5a734204cc6891c34586bd62243f8421@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@157.161.10.35>
Content-Length: 0


<------------>
   -- Executing [0901456056@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@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@vl-xm001001.imp.ch;tag=5a20ba5a28f8e172062f83d711ed55cf
To: <sip:0901456056@157.161.10.35;user=phone>;tag=as1017af28
Call-ID: 5a734204cc6891c34586bd62243f8421@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@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@from-alcatel:3] Macro("SIP/alcatel-0000aad1", "dundi,41901456056") in new stack
   -- Executing [s@macro-dundi:1] NoOp("SIP/alcatel-0000aad1", "DUNDi Lookup 41901456056") in new stack
   -- Executing [s@macro-dundi:2] Goto("SIP/alcatel-0000aad1", "41901456056,1") in new stack
   -- Goto (macro-dundi,41901456056,1)
   -- Executing [0901456056@from-alcatel:4] Dial("SIP/alcatel-0000aad1", "SIP/41901456056@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@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@157.161.10.35>;tag=as520dcb3b
To: <sip:41901456056@157.161.10.10>
Contact: <sip:41618269314@157.161.10.35>
Call-ID: 1c4ad0e32b89deb832dce3dd4bc50813@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@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@157.161.10.35>;tag=as520dcb3b
To: <sip:41901456056@157.161.10.10>
Call-ID: 1c4ad0e32b89deb832dce3dd4bc50813@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@157.161.10.35>;tag=as520dcb3b
To: <sip:41901456056@157.161.10.10>;tag=f1f26313
Call-ID: 1c4ad0e32b89deb832dce3dd4bc50813@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@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@157.161.3.19
To: <sip:0901456056@157.161.10.35;user=phone>
CSeq: 1349603881 CANCEL
From: sip:0618269314@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@vl-xm001001.imp.ch;tag=5a20ba5a28f8e172062f83d711ed55cf
To: <sip:0901456056@157.161.10.35;user=phone>;tag=as1017af28
Call-ID: 5a734204cc6891c34586bd62243f8421@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@vl-xm001001.imp.ch;tag=5a20ba5a28f8e172062f83d711ed55cf
To: <sip:0901456056@157.161.10.35;user=phone>;tag=as1017af28
Call-ID: 5a734204cc6891c34586bd62243f8421@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@157.161.10.35' in 32000 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 157.161.10.10:5060:
CANCEL sip:41901456056@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@157.161.10.35>;tag=as520dcb3b
To: <sip:41901456056@157.161.10.10>
Call-ID: 1c4ad0e32b89deb832dce3dd4bc50813@157.161.10.35
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.2.15
Content-Length: 0


---
Scheduling destruction of SIP dialog '1c4ad0e32b89deb832dce3dd4bc50813@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@157.161.10.35>;tag=as520dcb3b
To: <sip:41901456056@157.161.10.10>
Call-ID: 1c4ad0e32b89deb832dce3dd4bc50813@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@157.161.10.35>;tag=as520dcb3b
To: <sip:41901456056@157.161.10.10>;tag=f1f26313
Call-ID: 1c4ad0e32b89deb832dce3dd4bc50813@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@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@157.161.10.35>;tag=as520dcb3b
To: <sip:41901456056@157.161.10.10>;tag=f1f26313
Contact: <sip:41618269314@157.161.10.35>
Call-ID: 1c4ad0e32b89deb832dce3dd4bc50813@157.161.10.35
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.15
Content-Length: 0


---
Really destroying SIP dialog '1c4ad0e32b89deb832dce3dd4bc50813@157.161.10.35' Method: INVITE

<--- SIP read from UDP:157.161.3.19:5060 --->
ACK sip:0901456056@157.161.10.35;user=phone SIP/2.0
Call-ID: 5a734204cc6891c34586bd62243f8421@157.161.3.19
From: sip:0618269314@vl-xm001001.imp.ch;tag=5a20ba5a28f8e172062f83d711ed55cf
To: <sip:0901456056@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) ---
By: Benoit IMP Panizzon (imp) 2011-02-14 02:33:20.000-0600

Best is to check sip2.pcap with the wireshark built-in voip functions which can show a graph of sip and rtp signalling and flows. So you see, that asterisk is receiving early media, but not passing it to the caller.
By: Benoit IMP Panizzon (imp) 2011-02-14 02:38:51.000-0600

Equipment used:

Asterisk 2.6.2.15 on FreeBSD as SIP Gateway to 'Trunk' Customers for screening and routing purposes which cannot be achived by the Carrier PBX.

Alcatel OmniPCX Enterprise R9.0 h1.301.40 as 'Trunk' Client atteched to Asterisk. But problem can also be reproduced with Grandstream, or ZyXEL V2000W or N900 Sip Client (and probaly many more, as the pcap shows asterisk is not sending rtp).

Carrier PBX: IPGallery Release: 4.1-15_SS (www.ipgallery.com).

PS: I did play around with sending 'Progressing' or not, setting 'progressinboand' to no, never, yes and  prematuremedia to 'no' or 'yes'. The resulty I got vary, but none of them cause asterisk to actualy forward the early audio it got from the carrier to the client. Asterisk either generated early audio 'ringing' itself or told the client to generate 'ringing'.
By: Russell Bryant (russell) 2011-07-26 14:28:07.133-0500

Per the Asterisk maintenance timeline page at http://www.asterisk.org/asterisk-versions maintenance (bug) support for the 1.4 and 1.6.x branches has ended. For continued maintenance support please move to the 1.8 branch which is a long term support (LTS) branch. For more information about branch support, please see https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

If this is still an issue, please open a new issue so it can be re-triaged appropriately. Thanks!