[Asterisk-Users] one way audio on oh323 channel, there's no rtp traffic

Fri Nov 4 06:33:02 MST 2005

Hi all,

i'm experiencing a one way call only between a ipPhone
and an analog one through a oh323 channel between 
my asterisk and a Nortel GK.

Doing some sniffing and some debug with ethereal and
tcpump i can say (i hope, as newby to say the right
thing) that i can't see any rtp traffic
between the asterisk and the nortel.
In the analog phone (in the outside telecom world) i
can't ear nothing said in the ipPhone.
Viceversa in the ipPhone (Mitel one) i can ear the
voice comming from the outside world.

In my sip.conf

[419]
callerid=0432281316 TEST <test 419>
type=friend
username=419
secret=password
host=dynamic
nat=yes
canreinvite=no
reinvite=no
disallow=all
allow=ulaw
allow=gsm
;allow=alaw
dtmfmode=rfc2833
context=out
callgroup=1
pickupgroup=1

There's no rtp traffic from the phone or from the
asterisk to the GK.
The GK stays on the intranet even if it has a internet
looking ip.

ipPhone 10.24.3.40
asterisk 10.24.2.253
GK 80.74.178.196

Issuing on asterisk rtp debug
[2]WrapH323EndPoint::AnswerCall: Request to answer
call ip$80.74.178.196:34404/1169
Got RTP packet from 10.24.3.40:20012 (type 0, seq 14,
ts -1120604096, len 160)
[2]WrapH323EndPoint::AnswerCall: Call answered
[ip$80.74.178.196:34404/1169]
Got RTP packet from 10.24.3.40:20012 (type 0, seq 15,
ts -1120603936, len 160)
Got RTP packet from 10.24.3.40:20012 (type 0, seq 16,
ts -1120603776, len 160)
[2]WrapH323Connection::OnReceivedFacility: Received
FACILITY message [ip$80.74.178.196:34404/1169]
[2]WrapH323Connection::OnReceivedFacility: Received
FACILITY message [ip$80.74.178.196:34404/1169]
[2]WrapH323Connection::OnReceivedFacility: Received
FACILITY message [ip$80.74.178.196:34404/1169]
[3]WrapH323EndPoint::OpenAudioChannel: Direction =>
RECODER, Buffer => 320
[2]WrapH323EndPoint::OpenAudioChannel: Media format:
FrameSize 8, FrameTime 8, TimeUnits 8
[2]WrapH323EndPoint::OpenAudioChannel: Codec info:
FrameRate 160
[2]WrapH323EndPoint::OpenAudioChannel: Packet size:
160
[2]WrapH323EndPoint::OpenAudioChannel: Frames per
packet: 20
[2]WrapH323EndPoint::OpenAudioChannel: LID Codec
G.711-uLaw-64k
[3]WrapH323EndPoint::OpenAudioChannel: The sound
channel with the application is asterisk-oh323(fd=42)
[4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
initialized.
[4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
initialized.
[4]PAsteriskSoundChannel::PAsteriskSoundChannel:
Object initialized.
[3]PAsteriskSoundChannel::Open: os_handle 42,
mediaFormat 0, frameTime 1 ms, frameNum 20, packetSize
160
[3]WrapH323EndPoint::OpenAudioChannel: Opened sound
channel "Asterisk" for recording using 1x320 byte
buffers.
[3]WrapH323Connection::OnEstablished:
WrapH323Connection [ip$80.74.178.196:34404/1169]
established (FastStartDisabled/H245Tunneling)
[3]WrapH323EndPoint::OnConnectionEstablished:
Connection [ip$80.74.178.196:34404/1169] established.
[3]WrapH323EndPoint::GetConnectionInfo:
[ip$80.74.178.196:34404/1169] RTP Media:
10.24.2.253:21002-0.0.0.0:0
Got RTP packet from 10.24.3.40:20012 (type 0, seq 17,
ts -1120603616, len 160)
Got RTP packet from 10.24.3.40:20012 (type 0, seq 18,
ts -1120603456, len 160)
[2]WrapH323Connection::OnReceivedFacility: Received
FACILITY message [ip$80.74.178.196:34404/1169]
Got RTP packet from 10.24.3.40:20012 (type 0, seq 19,
ts -1120603296, len 160)
[3]WrapH323EndPoint::OpenAudioChannel: Direction =>
PLAYER, Buffer => 320
[2]WrapH323EndPoint::OpenAudioChannel: Media format:
FrameSize 8, FrameTime 8, TimeUnits 8
[2]WrapH323EndPoint::OpenAudioChannel: Codec info:
FrameRate 160
[2]WrapH323EndPoint::OpenAudioChannel: Packet size:
160
[2]WrapH323EndPoint::OpenAudioChannel: Frames per
packet: 20
[2]WrapH323EndPoint::OpenAudioChannel: LID Codec
G.711-uLaw-64k
[3]WrapH323EndPoint::OpenAudioChannel: The sound
channel with the application is asterisk-oh323(fd=40)
[4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
initialized.
[4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
initialized.
[4]PAsteriskSoundChannel::PAsteriskSoundChannel:
Object initialized.
[3]PAsteriskSoundChannel::Open: os_handle 40,
mediaFormat 0, frameTime 1 ms, frameNum 20, packetSize
160
[3]WrapH323EndPoint::OpenAudioChannel: Opened sound
channel "Asterisk" for playing using 1x320 byte
buffers.
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
Sent RTP packet to 10.24.3.40:20012 (type 0, seq
26203, ts 160, len 160)
[2]WrapH323Connection::OnReceivedFacility: Received
FACILITY message [ip$80.74.178.196:34404/1169]
[5]PAsteriskSoundChannel::Read: Data read [320 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
Got RTP packet from 10.24.3.40:20012 (type 0, seq 20,
ts -1120603136, len 160)
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
Sent RTP packet to 10.24.3.40:20012 (type 0, seq
26204, ts 320, len 160)
[5]PAsteriskSoundChannel::Read: Data read [320 bytes]
Got RTP packet from 10.24.3.40:20012 (type 0, seq 21,
ts -1120602976, len 160)
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
Sent RTP packet to 10.24.3.40:20012 (type 0, seq
26205, ts 480, len 160)
Got RTP packet from 10.24.3.40:20012 (type 0, seq 22,
ts -1120602816, len 160)
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
Sent RTP packet to 10.24.3.40:20012 (type 0, seq
26206, ts 640, len 160)
[5]PAsteriskSoundChannel::Read: Data read [320 bytes]
Got RTP packet from 10.24.3.40:20012 (type 0, seq 23,
ts -1120602656, len 160)
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
Sent RTP packet to 10.24.3.40:20012 (type 0, seq
26207, ts 800, len 160)
Got RTP packet from 10.24.3.40:20012 (type 0, seq 24,
ts -1120602496, len 160)
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
Sent RTP packet to 10.24.3.40:20012 (type 0, seq
26208, ts 960, len 160)

snip
[5]PAsteriskSoundChannel::Read: Data read [320 bytes]
Got RTP packet from 10.24.3.40:20012 (type 0, seq 45,
ts -1120599136, len 160)
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
Sent RTP packet to 10.24.3.40:20012 (type 0, seq
26229, ts 4320, len 160)
Got RTP packet from 10.24.3.40:20012 (type 0, seq 46,
ts -1120598976, len 160)
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
Sent RTP packet to 10.24.3.40:20012 (type 0, seq
26230, ts 4480, len 160)
[5]PAsteriskSoundChannel::Read: Data read [320 bytes]
Got RTP packet from 10.24.3.40:20012 (type 0, seq 47,
ts -1120598816, len 160)
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
Sent RTP packet to 10.24.3.40:20012 (type 0, seq
26231, ts 4640, len 160)
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
Sent RTP packet to 10.24.3.40:20012 (type 0, seq
26232, ts 4800, len 160)
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
Sent RTP packet to 10.24.3.40:20012 (type 0, seq
26233, ts 4960, len 160)
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
Sent RTP packet to 10.24.3.40:20012 (type 0, seq
26234, ts 5120, len 160)
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]

snip
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[3]WrapH323EndPoint::SetClearCallCause: Setting the
Q.931 cause code of connection
[ip$80.74.178.196:34404/1169], at 16
[2]WrapperAPI::h323_clear_call: Clearing call.
[4]ClearCallThread::ClearCallThread: Object
initialized.
[4]ClearCallThread::ClearCallThread: Unblock pipe -
45, 46
[2]WrapH323EndPoint::ClearCall: Request to clear call
[ip$80.74.178.196:34404/1169]
[2]WrapH323Connection::OnSendReleaseComplete: Sending
RELEASE COMPLETE message [ip$80.74.178.196:34404/1169]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[2]ClearCallThread::Main: Call with token
ip$80.74.178.196:34404/1169 cleared.
[4]ClearCallThread::ClearCallThread: Object deleted.
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[3]PAsteriskSoundChannel::Close: Closing os_handle 40
[3]PAsteriskSoundChannel::Close: Closing os_handle 42
[3]PAsteriskSoundChannel::PAsteriskSoundChannel: Total
I/Os: read=0, write=62
[3]PAsteriskSoundChannel::PAsteriskSoundChannel: Short
I/Os: write=0
[4]PAsteriskSoundChannel::PAsteriskSoundChannel:
Object deleted.
[4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
deleted.
[4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
deleted.
[3]PAsteriskSoundChannel::PAsteriskSoundChannel: Total
I/Os: read=64, write=0
[3]PAsteriskSoundChannel::PAsteriskSoundChannel: Short
I/Os: write=0
[4]PAsteriskSoundChannel::PAsteriskSoundChannel:
Object deleted.
[4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
deleted.
[4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
deleted.
[2]WrapH323EndPoint::ClearCall: Request to clear call
[ip$80.74.178.196:34404/1169]
[2]WrapH323EndPoint::OnConnectionCleared: Connection
[ip$80.74.178.196:34404/1169] closed.
[2]WrapH323EndPoint::OnConnectionCleared: Call with
"E164:0432707350 [80.74.178.196]" completed
[4]WrapH323Connection::WrapH323Connection:
WrapH323Connection deleted.

I can't see rtp traffic from asterisk to the gk !

Today i've changed the rtp.conf and oh323.conf file to
tell my asterisk to use rtp ports from 21000 to 30000.
issuing oh323 show conf 
 Configuration of OpenH323 channel driver
------------------------------------------
Version: 0.7.3
Listening on address: 10.24.2.253:1720
Gatekeeper used: Nortel_H323_Gatekeeper at 80.74.178.196
(Registered)
FastStart/H245Tunnelling/H245inSetup: OFF/ON/ON
Supported formats in pref. order: ulaw<0>
Jitter buffer limits (min/max): 20-100 ms
TCP port range: 10000 - 20000
UDP (RAS) port range: 10000 - 20000
UDP (RTP) port range: 21000 - 30000
IP Type-of-Service value: 0
User input mode: rfc2833
Max number of inbound H.323 calls: 100
Max number of outbound H.323 calls: 100
Max number of simultaneous H.323 calls: 100
Max call rate (ingress direction): 1.00/30
Default language: en
Default music class: default
Default context: voip-h323

and during the test i can see issuing
tcpdump -i eth0 -n host 80.74.178.196

snip

17:23:14.441390 IP 80.74.178.196.34396 >
10.24.2.253.1720: P 852:913(61) ack 794 win 10887
17:23:14.442622 IP 10.24.2.253.1720 >
80.74.178.196.34396: F 794:794(0) ack 913 win 6432
17:23:14.469007 IP 80.74.178.196.34396 >
10.24.2.253.1720: . ack 795 win 0
17:23:14.469416 IP 80.74.178.196.34396 >
10.24.2.253.1720: F 913:913(0) ack 795 win 0
17:23:14.469435 IP 10.24.2.253.1720 >
80.74.178.196.34396: . ack 914 win 6432
17:23:14.470624 IP 10.24.2.253.10002 >
80.74.178.196.1719: UDP, length: 175
17:23:14.501858 IP 80.74.178.196.1719 >
10.24.2.253.10002: UDP, length: 3
17:23:38.970892 IP 10.24.2.253.10002 >
80.74.178.196.1719: UDP, length: 372
17:23:39.011764 IP 80.74.178.196.1719 >
10.24.2.253.10002: UDP, length: 137
17:24:04.015093 IP 10.24.2.253.10002 >
80.74.178.196.1719: UDP, length: 372
17:24:04.058344 IP 80.74.178.196.1719 >
10.24.2.253.10002: UDP, length: 137
17:24:29.061334 IP 10.24.2.253.10002 >
80.74.178.196.1719: UDP, length: 372
17:24:29.106223 IP 80.74.178.196.1719 >
10.24.2.253.10002: UDP, length: 137

no traffic from ports in the range from 21000 to 30000
is made !!!

ROUTING AND FIREWALL
The telco also handles my wan and they say me that
there're no firewall/routers that drop the traffic.
I don't really trust very much in them ! 

CODECS
The phone, asterisk and the gk all use 
ulaw   (G.711 u-law)

Other sip calls between sip phones and sip and analog
(locallly configured zaptel interfaces) are normally
working on the same asterisk
Any idea ?

Mik

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