[asterisk-bugs] [JIRA] (ASTERISK-27763) PJSIP doesn't accept a call when INVITE contains T.38 stream, chan_sip does
Thiago Coutinho (JIRA)
noreply at issues.asterisk.org
Thu Mar 22 12:19:39 CDT 2018
Thiago Coutinho created ASTERISK-27763:
------------------------------------------
Summary: PJSIP doesn't accept a call when INVITE contains T.38 stream, chan_sip does
Key: ASTERISK-27763
URL: https://issues.asterisk.org/jira/browse/ASTERISK-27763
Project: Asterisk
Issue Type: Bug
Security Level: None
Components: pjproject/pjsip
Affects Versions: 13.20.0
Environment: CentOS Linux release 7.4.1708 (Core)
Kernel 3.10.0-693.17.1.el7.x86_64
Asterisk 13.20.0
Reporter: Thiago Coutinho
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.
pjsip.conf:
```
[voxip]
type=registration
outbound_auth=voxip
server_uri=sip:10.150.129.68
client_uri=sip:4730863277 at 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
```
pjsip trace:
```
SIP ->
Request
INVITE sip:4731215050 at 10.143.92.98:5060;transport=UDP;user=phone SIP/2.0
From:<sip:11992567632 at 10.150.129.68:5060;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56
To:<sip:4731215050 at 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 at 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 at 10.143.92.98:5060;transport=UDP;user=phone SIP/2.0
From:<sip:11992567632 at 10.150.129.68:5060;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56
To:<sip:4731215050 at 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 at 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 at 10.150.129.68;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56
To:<sip:4731215050 at 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 at 10.150.129.68;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56
To:<sip:4731215050 at 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 at 10.143.92.98:5060;transport=UDP;user=phone SIP/2.0
From:<sip:11992567632 at 10.150.129.68:5060;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56
To:<sip:4731215050 at 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 at 10.143.92.98:5060;transport=UDP;user=phone SIP/2.0
From:<sip:11992567632 at 10.150.129.68:5060;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56
To:<sip:4731215050 at 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.conf:
```
[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
```
chan_sip trace:
```
<--- SIP read from UDP:10.150.129.68:5060 --->
INVITE sip:4731215050 at 10.143.92.98:5060;transport=UDP;user=phone SIP/2.0
From: <sip:11987291094 at 10.150.129.68:5060;user=phone>;tag=-13c4-45026-1c4bf-8b2a32c-1c4bf
To: <sip:4731215050 at 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 at 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 at 10.150.129.68:5060;user=phone>;tag=-13c4-45026-1c4bf-8b2a32c-1c4bf
To: <sip:4731215050 at 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 at 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 at 10.150.129.68:5060;user=phone>;tag=-13c4-45026-1c4bf-8b2a32c-1c4bf
To: <sip:4731215050 at 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 at 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 at 10.143.92.98:5060 SIP/2.0
From: <sip:11987291094 at 10.150.129.68:5060;user=phone>;tag=-13c4-45026-1c4bf-8b2a32c-1c4bf
To: <sip:4731215050 at 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
<------------->
```
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