[Home]

Summary:ASTERISK-27763: res_pjsip_session: Initial INVITE with audio+fax results in 488 instead of declining stream
Reporter:Thiago Coutinho (thiagocnet)Labels:fax pjsip
Date Opened:2018-03-22 12:19:13Date Closed:2018-07-09 05:19:09
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Resources/res_pjsip_session
Versions:13.20.0 Frequency of
Occurrence		Constant
Related
Issues:
Environment:CentOS Linux release 7.4.1708 (Core) Kernel 3.10.0-693.17.1.el7.x86_64 Asterisk 13.20.0Attachments:
Description:Some providers send T.38 streams along with the call (I don't know why) causing PJSIP to reject the call. chan_sip on the other hand accepts the call normally.

{code:title=pjsip.conf|borderStyle=solid}
[voxip]
type=registration
outbound_auth=voxip
server_uri=sip:10.150.129.68
client_uri=sip:4730863277@10.150.129.68
auth_rejection_permanent=no

[voxip]
type=auth
auth_type=userpass
username=4730863277
password=4730863277

[voxip]
type=aor
contact=sip:10.150.129.68
qualify_frequency=60

[voxip]
type=endpoint
context=from-pstn
allow=!all,g729,alaw
;auth=voxip
outbound_auth=voxip
aors=voxip
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
from_user=4730863277
from_domain=10.150.129.68
t38_udptl=yes
t38_udptl_ec=redundancy
fax_detect=no
t38_udptl_nat=yes

[voxip]
type=identify
endpoint=voxip
match=10.150.129.68
{code}

{code:title=pjsip trace|borderStyle=solid}
SIP ->

    Request
    INVITE sip:4731215050@10.143.92.98:5060;transport=UDP;user=phone SIP/2.0
    From:<sip:11992567632@10.150.129.68:5060;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56
    To:<sip:4731215050@10.143.92.98:5060;user=phone>
    Call-ID:eb76d2304481960a13c47f4a56f13a6220a1b439c2a972a640-0008-7943
    CSeq: 1 INVITE
    User-agent:CS2000_NGSS/9.0
    P-Asserted-Identity:<sip:11992567632@10.150.129.68;user=phone>
    Allow:ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
    Via:SIP/2.0/UDP SOO2CS2K:5060;maddr=10.150.129.68;branch=z9hG4bK-7f4a56-f13a6220-5c9ab8e5
    Max-Forwards:140
    Contact:<sip:10.150.129.68:5060;transport=UDP>
    Supported:100rel,resource-priority
    Content-Type: application/sdp
    Content-Length:420

SDP ->

    Version = 0.
    Owner = PVG 1521481511010 1521481511010 IN IP4 10.152.205.107.
    Session Name = -.
    Phone Address = +1 6135555555.
    Connection = IN IP4 10.152.205.107.
    Time = 0 0.
    Media Name = audio 56534 RTP/AVP 18 8 101.
    Media Attribute = rtpmap:101 telephone-event/8000.
    Media Attribute = a=fmtp:101 0-15.
    Media Attribute = a=ptime:20.
    Media Attribute = a=fmtp:18 annexb=no.
    Media Attribute = m=image 64726 udptl t38.
    Media Attribute = a=T38FaxVersion:0.
    Media Attribute = a=T38FaxMaxBuffer:1100.
    Media Attribute = a=T38FaxMaxDatagram:612.
    Media Attribute = a=T38MaxBitRate:14400.
    Media Attribute = a=T38FaxRateManagement:transferredTCF.
    Media Attribute = a=T38FaxUdpEC:t38UDPRedundancy.

SIP ->

    Request
    INVITE sip:4731215050@10.143.92.98:5060;transport=UDP;user=phone SIP/2.0
    From:<sip:11992567632@10.150.129.68:5060;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56
    To:<sip:4731215050@10.143.92.98:5060;user=phone>
    Call-ID:eb76d2304481960a13c47f4a56f13a6220a1b439c2a972a640-0008-7943
    CSeq: 1 INVITE
    User-agent:CS2000_NGSS/9.0
    P-Asserted-Identity:<sip:11992567632@10.150.129.68;user=phone>
    Allow:ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
    Via:SIP/2.0/UDP SOO2CS2K:5060;maddr=10.150.129.68;branch=z9hG4bK-7f4a56-f13a6220-5c9ab8e5
    Max-Forwards:140
    Contact:<sip:10.150.129.68:5060;transport=UDP>
    Supported:100rel,resource-priority
    Content-Type: application/sdp
    Content-Length:420

SDP ->

    Version = 0.
    Owner = PVG 1521481511010 1521481511010 IN IP4 10.152.205.107.
    Session Name = -.
    Phone Address = +1 6135555555.
    Connection = IN IP4 10.152.205.107.
    Time = 0 0.
    Media Name = audio 56534 RTP/AVP 18 8 101.
    Media Attribute = rtpmap:101 telephone-event/8000.
    Media Attribute = a=fmtp:101 0-15.
    Media Attribute = a=ptime:20.
    Media Attribute = a=fmtp:18 annexb=no.
    Media Attribute = m=image 64726 udptl t38.
    Media Attribute = a=T38FaxVersion:0.
    Media Attribute = a=T38FaxMaxBuffer:1100.
    Media Attribute = a=T38FaxMaxDatagram:612.
    Media Attribute = a=T38MaxBitRate:14400.
    Media Attribute = a=T38FaxRateManagement:transferredTCF.
    Media Attribute = a=T38FaxUdpEC:t38UDPRedundancy.

SIP <-

    Response
    SIP/2.0 488 Not Acceptable Here
    Via:SIP/2.0/UDP SOO2CS2K:5060;rport=5060;maddr=10.150.129.68;received=10.150.129.68;branch=z9hG4bK-7f4a56-f13a6220-5c9ab8e5
    Call-ID:eb76d2304481960a13c47f4a56f13a6220a1b439c2a972a640-0008-7943
    From:<sip:11992567632@10.150.129.68;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56
    To:<sip:4731215050@10.143.92.98;user=phone>;tag=9f804ff5-215a-4c91-a93e-52e4689f866c
    CSeq: 1 INVITE
    Server:Asterisk PBX certified/13.13-cert7
    Content-Length:0

SIP <-

    Response
    SIP/2.0 488 Not Acceptable Here
    Via:SIP/2.0/UDP SOO2CS2K:5060;rport=5060;maddr=10.150.129.68;received=10.150.129.68;branch=z9hG4bK-7f4a56-f13a6220-5c9ab8e5
    Call-ID:eb76d2304481960a13c47f4a56f13a6220a1b439c2a972a640-0008-7943
    From:<sip:11992567632@10.150.129.68;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56
    To:<sip:4731215050@10.143.92.98;user=phone>;tag=9f804ff5-215a-4c91-a93e-52e4689f866c
    CSeq: 1 INVITE
    Server:Asterisk PBX certified/13.13-cert7
    Content-Length:0

SIP ->

    Request
    ACK sip:4731215050@10.143.92.98:5060;transport=UDP;user=phone SIP/2.0
    From:<sip:11992567632@10.150.129.68:5060;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56
    To:<sip:4731215050@10.143.92.98;user=phone>;tag=9f804ff5-215a-4c91-a93e-52e4689f866c
    Call-ID:eb76d2304481960a13c47f4a56f13a6220a1b439c2a972a640-0008-7943
    CSeq: 1 ACK
    User-agent:CS2000_NGSS/9.0
    Max-Forwards:70
    Allow:ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
    Via:SIP/2.0/UDP SOO2CS2K:5060;maddr=10.150.129.68;branch=z9hG4bK-7f4a56-f13a6220-5c9ab8e5
    Contact:<sip:10.150.129.68:5060;transport=UDP>
    Supported:100rel,resource-priority
    Content-Length:0

SIP <-

    Request
    ACK sip:4731215050@10.143.92.98:5060;transport=UDP;user=phone SIP/2.0
    From:<sip:11992567632@10.150.129.68:5060;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56
    To:<sip:4731215050@10.143.92.98;user=phone>;tag=9f804ff5-215a-4c91-a93e-52e4689f866c
    Call-ID:eb76d2304481960a13c47f4a56f13a6220a1b439c2a972a640-0008-7943
    CSeq: 1 ACK
    User-agent:CS2000_NGSS/9.0
    Max-Forwards:70
    Allow:ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
    Via:SIP/2.0/UDP SOO2CS2K:5060;maddr=10.150.129.68;branch=z9hG4bK-7f4a56-f13a6220-5c9ab8e5
    Contact:<sip:10.150.129.68:5060;transport=UDP>
    Supported:100rel,resource-priority
    Content-Length:0
{code}

{code:title=sip.conf|borderStyle=solid}
[voxip]
type=peer
defaultuser=4730863277
secret=4730863277
fromuser=4730863277
fromdomain=gvt.com.br
domain=gvt.com.br
host=10.150.129.68
context=from-pstn
dtmfmode=rfc2833
insecure=port,invite
qualify=yes
canreinvite=no
disallow=all
allow=alaw
nat=no
port=5060
ignoresdpversion=yes
busydetect=yes
busycount=3
t38pt_udptl=yes
{code}

{code:title=chan_sip trace|borderStyle=solid}
<--- SIP read from UDP:10.150.129.68:5060 --->
INVITE sip:4731215050@10.143.92.98:5060;transport=UDP;user=phone SIP/2.0
From: <sip:11987291094@10.150.129.68:5060;user=phone>;tag=-13c4-45026-1c4bf-8b2a32c-1c4bf
To: <sip:4731215050@10.143.92.98:5060;user=phone>
Call-ID: 7f9af3004481960a13c41c4bf6e88c5b4097a7dd504d8f40-0058-4883
CSeq: 1 INVITE
User-agent: CS2000_NGSS/9.0
P-Asserted-Identity: <sip:11987291094@10.150.129.68;user=phone>
Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
Via: SIP/2.0/UDP SOO2CS2K:5060;maddr=10.150.129.68;branch=z9hG4bK-1c4bf-6e88c5b-43954845
Max-Forwards: 140
Contact: <sip:10.150.129.68:5060;transport=UDP>
Supported: 100rel,resource-priority
Content-Type: application/sdp
Content-Length: 418

v=0
o=PVG 1521732832740 1521732832740 IN IP4 10.152.204.43
s=-
p=+1 6135555555
c=IN IP4 10.152.204.43
t=0 0
m=audio 49330 RTP/AVP 18 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=fmtp:18 annexb=no
m=image 57522 udptl t38
a=T38FaxVersion:0
a=T38FaxMaxBuffer:1100
a=T38FaxMaxDatagram:612
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (14 headers 18 lines) ---
Sending to 10.150.129.68:5060 (NAT)
Sending to 10.150.129.68:5060 (NAT)
Using INVITE request as basis request - 7f9af3004481960a13c41c4bf6e88c5b4097a7dd504d8f40-0058-4883
Found peer 'VOXIP_GVT' for '11987291094' from 10.150.129.68:5060
 == Using SIP RTP CoS mark 5
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
 == Using UDPTL CoS mark 5                                                                                                                                                                  [107/1736]
Got T.38 offer in SDP in dialog 7f9af3004481960a13c41c4bf6e88c5b4097a7dd504d8f40-0058-4883
Capabilities: us - (alaw), peer - audio=(alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.152.204.43:49330
Looking for 4731215050 in from-pstn (domain 10.143.92.98)
sip_route_dump: route/path hop: <sip:10.150.129.68:5060;transport=UDP>

<--- Transmitting (no NAT) to 10.150.129.68:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP SOO2CS2K:5060;maddr=10.150.129.68;branch=z9hG4bK-1c4bf-6e88c5b-43954845;received=10.150.129.68
From: <sip:11987291094@10.150.129.68:5060;user=phone>;tag=-13c4-45026-1c4bf-8b2a32c-1c4bf
To: <sip:4731215050@10.143.92.98:5060;user=phone>
Call-ID: 7f9af3004481960a13c41c4bf6e88c5b4097a7dd504d8f40-0058-4883
CSeq: 1 INVITE
Server: Asterisk PBX 13.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:4731215050@10.143.92.98:5060>
Content-Length: 0


<------------>
Audio is at 14648
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 10.150.129.68:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP SOO2CS2K:5060;maddr=10.150.129.68;branch=z9hG4bK-1c4bf-6e88c5b-43954845;received=10.150.129.68
From: <sip:11987291094@10.150.129.68:5060;user=phone>;tag=-13c4-45026-1c4bf-8b2a32c-1c4bf
To: <sip:4731215050@10.143.92.98:5060;user=phone>;tag=as2030a2ce
Call-ID: 7f9af3004481960a13c41c4bf6e88c5b4097a7dd504d8f40-0058-4883
CSeq: 1 INVITE
Server: Asterisk PBX 13.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:4731215050@10.143.92.98:5060>
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 413021425 413021425 IN IP4 10.143.92.98                                                                                                                                                [61/1736]
s=Asterisk PBX 13.20.0
c=IN IP4 10.143.92.98
t=0 0
m=audio 14648 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=image 0 udptl t38

<------------>

<--- SIP read from UDP:10.150.129.68:5060 --->
ACK sip:4731215050@10.143.92.98:5060 SIP/2.0
From: <sip:11987291094@10.150.129.68:5060;user=phone>;tag=-13c4-45026-1c4bf-8b2a32c-1c4bf
To: <sip:4731215050@10.143.92.98:5060;user=phone>;tag=as2030a2ce
Call-ID: 7f9af3004481960a13c41c4bf6e88c5b4097a7dd504d8f40-0058-4883
CSeq: 1 ACK
User-agent: CS2000_NGSS/9.0
Max-Forwards: 70
Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
Via: SIP/2.0/UDP SOO2CS2K:5060;maddr=10.150.129.68;branch=z9hG4bK-1c4bf-6e88c76-6ec0f83c
Contact: <sip:10.150.129.68:5060;transport=UDP>
Supported: 100rel,resource-priority
Content-Length: 0

<------------->
{code}
Comments:By: Asterisk Team (asteriskteam) 2018-03-22 12:19:15.974-0500

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: Friendly Automation (friendly-automation) 2018-07-09 05:19:12.396-0500

Change 9352 merged by Joshua Colp:
res_pjsip_t38: Decline T.38 stream on failure case.

[https://gerrit.asterisk.org/9352|https://gerrit.asterisk.org/9352]
By: Friendly Automation (friendly-automation) 2018-07-09 05:34:14.743-0500

Change 9354 merged by Joshua Colp:
res_pjsip_t38: Decline T.38 stream on failure case.

[https://gerrit.asterisk.org/9354|https://gerrit.asterisk.org/9354]
By: Friendly Automation (friendly-automation) 2018-07-09 05:39:20.518-0500

Change 9353 merged by Joshua Colp:
res_pjsip_t38: Decline T.38 stream on failure case.

[https://gerrit.asterisk.org/9353|https://gerrit.asterisk.org/9353]
By: Thiago Coutinho (thiagocnet) 2018-07-09 06:38:29.576-0500

Hi Joshua. With this change the call will be accepted (like in chan_sip) or declined?
By: Asterisk Team (asteriskteam) 2018-07-09 06:38:29.845-0500

This issue has been reopened as a result of your commenting on it as the reporter. It will be triaged once again as applicable.
By: Joshua C. Colp (jcolp) 2018-07-09 06:42:13.375-0500

The audio portion is accepted. The fax portion is declined.