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Summary:ASTERISK-26769: T38 informations during session progress are not passed to the client
Reporter:Federico Santulli (fsantulli)Labels:
Date Opened:2017-02-06 06:40:40.000-0600Date Closed:2017-02-06 06:54:49.000-0600
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_pjsip
Versions:13.9.1 Frequency of
Occurrence		Constant
Related
Issues:
Environment:Linux 4.4.39Attachments:
Description:We are sending an invite with early media enabled. During the 183 Session Progress we do receive T38 availability information from the peer. These information are not sent back to the user registered on asterisk.

Here is the SIP TRACE:

U 2017/02/06 11:46:32.792320 10.10.10.23:5060 -> 10.10.10.6:5060
INVITE sip:014436888@10.10.10.6 SIP/2.0.
Via: SIP/2.0/UDP 10.10.10.23:5060;rport;branch=z9hG4bKPj9865ec86-cf71-43fb-9824-87434a978fe1.
From: <sip:014429891@10.10.10.23>;tag=cc596c91-d59b-4372-9f5c-496917abb395.
To: <sip:014436888@10.10.10.6>.
Contact: <sip:asterisk@10.10.10.23:5060>.
Call-ID: f917dd62-1eb7-4d8d-a284-8e5a123cae01.
CSeq: 29 INVITE.
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE, REGISTER.
Supported: 100rel, timer, replaces, norefersub.
Session-Expires: 1800.
Min-SE: 90.
P-Asserted-Identity: <sip:014429891@localhost;user=phone>.
Remote-Party-ID: <sip:014429891@localhost;user=phone>; party=calling; privacy=off; screen=yes.
Max-Forwards: 70.
User-Agent: Asterisk PBX v13.9.1.
Content-Type: application/sdp.
Content-Length:   230.
.
v=0.
o=- 245657452 245657452 IN IP4 10.10.10.23.
s=-.
c=IN IP4 10.10.10.23.
t=0 0.
m=audio 10400 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=maxptime:150.
a=sendrecv.


U 2017/02/06 11:46:32.806085 10.10.10.6:5060 -> 10.10.10.23:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 10.10.10.23:5060;received=10.10.10.23;rport=5060;branch=z9hG4bKPj9865ec86-cf71-43fb-9824-87434a978fe1.
From: <sip:014429891@10.10.10.23>;tag=cc596c91-d59b-4372-9f5c-496917abb395.
To: <sip:014436888@10.10.10.6>.
Call-ID: f917dd62-1eb7-4d8d-a284-8e5a123cae01.
CSeq: 29 INVITE.
Content-Length: 0.
.


U 2017/02/06 11:46:34.903827 10.10.10.6:5060 -> 10.10.10.23:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP 10.10.10.23:5060;received=10.10.10.23;rport=5060;branch=z9hG4bKPj9865ec86-cf71-43fb-9824-87434a978fe1.
From: <sip:014429891@10.10.10.23>;tag=cc596c91-d59b-4372-9f5c-496917abb395.
To: <sip:014436888@10.10.10.6>;tag=SDngma399-3299127679723201726114632.
Call-ID: f917dd62-1eb7-4d8d-a284-8e5a123cae01.
CSeq: 29 INVITE.
Server: CS2000_NGSS/8.0.
Require: 100rel.
Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK.
RSeq: 654.
Contact: <sip:10.10.10.6:5060;transport=UDP>.
Supported: 100rel,resource-priority.
Content-Type: application/sdp.
Content-Length: 309.
.
v=0.
o=PVG 1486376546850 1486376546850 IN IP4 10.10.10.6.
s=-.
p=+1 6135555555.
c=IN IP4 10.10.10.6.
t=0 0.
a=sqn: 0.
a=cdsc: 1 image udptl t38.
a=cpar: a=T38FaxVersion:0.
a=cpar: a=T38FaxUdpEC:t38UDPRedundancy.
m=audio 26884 RTP/AVP 8 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20.


U 2017/02/06 11:46:34.904938 10.10.10.23:5060 -> 10.10.10.6:5060
PRACK sip:10.10.10.6:5060;transport=UDP SIP/2.0.
Via: SIP/2.0/UDP 10.10.10.23:5060;rport;branch=z9hG4bKPj1f82b815-81d1-41d3-bc53-de6459fa185c.
From: <sip:014429891@10.10.10.23>;tag=cc596c91-d59b-4372-9f5c-496917abb395.
To: <sip:014436888@10.10.10.6>;tag=SDngma399-3299127679723201726114632.
Call-ID: f917dd62-1eb7-4d8d-a284-8e5a123cae01.
CSeq: 30 PRACK.
RAck: 654 29 INVITE.
Max-Forwards: 70.
User-Agent: Asterisk PBX v13.9.1.
Content-Length:  0.
.


U 2017/02/06 11:46:34.906356 10.10.10.23:5060 -> 185.137.84.161:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP 185.137.84.161:5060;rport=5060;received=185.137.84.161;branch=z9hG4bK-69fc15a3.
Call-ID: d758fb81-fdb64dff@185.137.84.161.
From: "0119896124" <sip:0119896124@voip.asterisk.it>;tag=81ded99717c42c7o0.
To: <sip:013184293@voip.asterisk.it>;tag=b474dedb-6ac5-40b1-a1f9-53220ab064c5.
CSeq: 102 INVITE.
Server: Asterisk PBX v13.9.1.
Contact: <sip:10.10.10.23:5060>.
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE, REGISTER.
P-Asserted-Identity: <sip:013184293@voip.asterisk.it>.
Remote-Party-ID: <sip:013184293@voip.asterisk.it>;privacy=off;screen=no.
Content-Type: application/sdp.
Content-Length:   235.
.
v=0.
o=- 37112077 37112079 IN IP4 10.10.10.23.
s=Asterisk.
c=IN IP4 10.10.10.23.
t=0 0.
m=audio 13214 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=maxptime:150.
a=sendrecv.


U 2017/02/06 11:46:34.922489 10.10.10.6:5060 -> 10.10.10.23:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 10.10.10.23:5060;received=10.10.10.23;rport=5060;branch=z9hG4bKPj1f82b815-81d1-41d3-bc53-de6459fa185c.
From: <sip:014429891@10.10.10.23>;tag=cc596c91-d59b-4372-9f5c-496917abb395.
To: <sip:014436888@10.10.10.6>;tag=SDngma399-3299127679723201726114632.
Call-ID: f917dd62-1eb7-4d8d-a284-8e5a123cae01.
CSeq: 30 PRACK.
Server: CS2000_NGSS/8.0.
Supported: 100rel,resource-priority.
Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK.
Content-Length: 0.
Comments:By: Asterisk Team (asteriskteam) 2017-02-06 06:40:42.434-0600

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].
By: Joshua C. Colp (jcolp) 2017-02-06 06:54:49.538-0600

The chan_pjsip module (and chan_sip) both do not have support for RFC3407[1] which is present in your SDP. Adding this would be a new feature. Additionally we do not currently support T.38 in early media at all. We only support a re-invite with T.38 after the session has been established.

[1] https://tools.ietf.org/html/rfc3407