[Asterisk-video] is this possible? asterisk -> sip voip provider -> gsm network -> 3g phone
yasin celik
yasin.celik at birsys.com
Wed Jan 27 08:37:16 CST 2010
Hi,
We want to make callfile based video delivery to 3g enabled cell phones.
After a week of googling we found some info and resources and tried to
success.
The call is placed and answered but it stucks in dialplan after executing
h324m_gw_answer(). Technical details are below. Is it possible to make such
a call from sip trunk to directly 3g enabled cell phone? If it is not is
there another way to do it without an isdn line? (like umts gsm gateway
perhaps?)
System config
-------------
Asterisk 1.4.7.1 with
10217-asterisk-unrestricted-digital-llc-11595-1.4.17.patch
Libpri 1.4.4 with 13055-libpri-1.4.4-llc-transmit-receive-patch-0.1.txt
patch
I successfully compiled below modules with instructions in
http://asterisk-party.net/index.php/Asterisk_Video_3G_FR
libh324m.so
codec_amr.so
app_h324m.so
app_mp4.so
app_rtsp.so
callfile
--------
Set: CHANNEL(transfercapability)=VIDEO
Set: CHANNEL(userinformationlayer1)=38
Channel: SIP/0090XXXXXXXXXX at voicetrading
Context: msg
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Extension: s
Priority: 1
cli show codecs
---------------
...
...
8192 (1 << 13) (0x2000) audio amr (AMR NB)
65536 (1 << 16) (0x10000) image jpeg (JPEG image)
131072 (1 << 17) (0x20000) image png (PNG image)
262144 (1 << 18) (0x40000) video h261 (H.261 Video)
524288 (1 << 19) (0x80000) video h263 (H.263 Video)
1048576 (1 << 20) (0x100000) video h263p (H.263+ Video)
2097152 (1 << 21) (0x200000) video h264 (H.264 Video)
sip.conf
--------
[general]
videosupport=yes
disallow=all
allow=h263p
allow=h263
allow=h264
allow=h261
allow=amr
allow=gsm
allow=alaw
allow=ulaw
extensions.conf
---------------
[msg]
exten => s,1,Answer
exten => s,2,NoOp(ul1=${CHANNEL(userinformationlayer1)})
exten => s,3,NoOp(ul2=${CHANNEL(transfercapability)})
exten => s,4,h324m_gw(sendvideo at 3g_video)
[3g_video]
exten => sendvideo,1,h324m_gw_answer()
exten => sendvideo,2,mp4play(/tmp/demo.3gp)
cli output
----------
-- Attempting call on SIP/00905357676979 at voicetrading for s at msg:1 (Retry
1)
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
> Channel SIP/voicetrading-0823a580 was answered.
-- Executing [s at msg:1] Answer("SIP/voicetrading-0823a580", "") in new
stack
-- Executing [s at msg:2] NoOp("SIP/voicetrading-0823a580", "ul1=38") in
new stack
-- Executing [s at msg:3] NoOp("SIP/voicetrading-0823a580", "ul2=VIDEO") in
new stack
-- Executing [s at msg:4] h324m_gw("SIP/voicetrading-0823a580",
"sendvideo at 3g_video") in new stack
-- Executing [sendvideo at 3g_video:1]
h324m_gw_answer("Local/sendvideo at 3g_video-157d,2", "") in new stack
-- Remote UNIX connection
-- Remote UNIX connection disconnected
== Spawn extension (msg, s, 4) exited non-zero on
'SIP/voicetrading-0823a580'
== Spawn extension (3g_video, sendvideo, 1) exited non-zero on
'Local/sendvideo at 3g_video-157d,2'
Really destroying SIP dialog
'72704e65251b53a94f8727b873c116c0 at 91.205.172.196' Method: BYE
sip trace
---------
#
U 2010/01/27 15:54:56.933028 91.205.172.196:5060 -> 194.120.0.198:5060
INVITE sip:0090XXXXXXXXXX at sip.voicetrading.com SIP/2.0
Via: SIP/2.0/UDP 91.205.172.196:5060;branch=z9hG4bK1b37c1ca;rport
From: "Unknown" <sip:Unknown at 91.205.172.196>;tag=as7a873e05
To: <sip:0090XXXXXXXXXX at sip.voicetrading.com>
Contact: <sip:Unknown at 91.205.172.196>
Call-ID: 505c973b7d2c06115e6862787ca1774a at 91.205.172.196
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 27 Jan 2010 13:54:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 372
v=0
o=root 2685 2685 IN IP4 91.205.172.196
s=session
c=IN IP4 91.205.172.196
b=CT:384
t=0 0
m=audio 10266 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 19806 RTP/AVP 34 103
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=sendrecv
#
U 2010/01/27 15:54:56.942796 194.120.0.198:5060 -> 91.205.172.196:5060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 91.205.172.196:5060;branch=z9hG4bK1b37c1ca;rport
From: "Unknown" <sip:Unknown at 91.205.172.196>;tag=as7a873e05
To: <sip:0090XXXXXXXXXX at sip.voicetrading.com>
Contact: sip:0090XXXXXXXXXX at 194.120.0.198:5060
Call-ID: 505c973b7d2c06115e6862787ca1774a at 91.205.172.196
CSeq: 102 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
WWW-Authenticate: Digest
realm="sipdiscount.com",nonce="3509265298",algorithm=MD5
Content-Length: 0
#
U 2010/01/27 15:54:56.943036 91.205.172.196:5060 -> 194.120.0.198:5060
ACK sip:0090XXXXXXXXXX at sip.voicetrading.com SIP/2.0
Via: SIP/2.0/UDP 91.205.172.196:5060;branch=z9hG4bK1b37c1ca;rport
From: "Unknown" <sip:Unknown at 91.205.172.196>;tag=as7a873e05
To: <sip:0090XXXXXXXXXX at sip.voicetrading.com>
Contact: <sip:Unknown at 91.205.172.196>
Call-ID: 505c973b7d2c06115e6862787ca1774a at 91.205.172.196
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
#
U 2010/01/27 15:54:56.943339 91.205.172.196:5060 -> 194.120.0.198:5060
INVITE sip:0090XXXXXXXXXX at sip.voicetrading.com SIP/2.0
Via: SIP/2.0/UDP 91.205.172.196:5060;branch=z9hG4bK3cadce12;rport
From: "Unknown" <sip:Unknown at 91.205.172.196>;tag=as7a873e05
To: <sip:0090XXXXXXXXXX at sip.voicetrading.com>
Contact: <sip:Unknown at 91.205.172.196>
Call-ID: 505c973b7d2c06115e6862787ca1774a at 91.205.172.196
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="username", realm="sipdiscount.com",
algorithm=MD5, uri="sip:0090XXXXXXXXXX at sip.voicetrading.com",
nonce="3509265298", response="feb2e7315123e783252748325b71446e", opaque=""
Date: Wed, 27 Jan 2010 13:54:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 372
v=0
o=root 2685 2686 IN IP4 91.205.172.196
s=session
c=IN IP4 91.205.172.196
b=CT:384
t=0 0
m=audio 10266 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 19806 RTP/AVP 34 103
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=sendrecv
#
U 2010/01/27 15:54:56.953906 194.120.0.198:5060 -> 91.205.172.196:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 91.205.172.196:5060;branch=z9hG4bK3cadce12;rport
From: "Unknown" <sip:Unknown at 91.205.172.196>;tag=as7a873e05
To: <sip:0090XXXXXXXXXX at sip.voicetrading.com>
Contact: sip:0090XXXXXXXXXX at 194.120.0.198:5060
Call-ID: 505c973b7d2c06115e6862787ca1774a at 91.205.172.196
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0
#
U 2010/01/27 15:54:57.033535 194.120.0.198:5060 -> 91.205.172.196:5060
SIP/2.0 183 Session progress
Via: SIP/2.0/UDP 91.205.172.196:5060;branch=z9hG4bK3cadce12;rport
From: "Unknown" <sip:Unknown at 91.205.172.196>;tag=as7a873e05
To:
<sip:0090XXXXXXXXXX at sip.voicetrading.com>;tag=c51710acc52b10ac4af9c8301bd806
b
Contact: sip:0090XXXXXXXXXX at 194.120.0.198:5060
Call-ID: 505c973b7d2c06115e6862787ca1774a at 91.205.172.196
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
Content-Length: 200
v=0
o=username 1264600385 1264600385 IN IP4 194.120.0.43
s=SIP Call
c=IN IP4 194.120.0.43
t=0 0
m=audio 25510 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
#
U 2010/01/27 15:55:02.855279 194.120.0.198:5060 -> 91.205.172.196:5060
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 91.205.172.196:5060;branch=z9hG4bK3cadce12;rport
From: "Unknown" <sip:Unknown at 91.205.172.196>;tag=as7a873e05
To:
<sip:0090XXXXXXXXXX at sip.voicetrading.com>;tag=c51710acc52b10ac4af9c8301bd806
b
Contact: sip:0090XXXXXXXXXX at 194.120.0.198:5060
Call-ID: 505c973b7d2c06115e6862787ca1774a at 91.205.172.196
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
Content-Length: 200
v=0
o=username 1264600391 1264600391 IN IP4 194.120.0.43
s=SIP Call
c=IN IP4 194.120.0.43
t=0 0
m=audio 25510 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
#
U 2010/01/27 15:55:02.855713 91.205.172.196:5060 -> 194.120.0.198:5060
ACK sip:0090XXXXXXXXXX at 194.120.0.198:5060 SIP/2.0
Via: SIP/2.0/UDP 91.205.172.196:5060;branch=z9hG4bK73a7c26f;rport
From: "Unknown" <sip:Unknown at 91.205.172.196>;tag=as7a873e05
To:
<sip:0090XXXXXXXXXX at sip.voicetrading.com>;tag=c51710acc52b10ac4af9c8301bd806
b
Contact: <sip:Unknown at 91.205.172.196>
Call-ID: 505c973b7d2c06115e6862787ca1774a at 91.205.172.196
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
#
U 2010/01/27 15:55:19.904614 194.120.0.198:5060 -> 91.205.172.196:5060
BYE sip:Unknown at 91.205.172.196 SIP/2.0
Via: SIP/2.0/UDP 194.120.0.198:5060;branch=z9hG4bK3cadce12
From:
<sip:0090XXXXXXXXXX at sip.voicetrading.com>;tag=c51710acc52b10ac4af9c8301bd806
b
To: "Unknown" <sip:Unknown at 91.205.172.196>;tag=as7a873e05
Contact: sip:0090XXXXXXXXXX at 194.120.0.198:5060
Call-ID: 505c973b7d2c06115e6862787ca1774a at 91.205.172.196
CSeq: 1 BYE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0
#
U 2010/01/27 15:55:19.904820 91.205.172.196:5060 -> 194.120.0.198:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP
194.120.0.198:5060;branch=z9hG4bK3cadce12;received=194.120.0.198
From:
<sip:0090XXXXXXXXXX at sip.voicetrading.com>;tag=c51710acc52b10ac4af9c8301bd806
b
To: "Unknown" <sip:Unknown at 91.205.172.196>;tag=as7a873e05
Call-ID: 505c973b7d2c06115e6862787ca1774a at 91.205.172.196
CSeq: 1 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:Unknown at 91.205.172.196>
Content-Length: 0
#
U 2010/01/27 15:55:20.865373 91.205.172.196:5060 -> 194.120.0.198:5060
OPTIONS sip:sip.voicetrading.com SIP/2.0
Via: SIP/2.0/UDP 91.205.172.196:5060;branch=z9hG4bK15310eaf;rport
From: "Unknown" <sip:Unknown at 91.205.172.196>;tag=as3f83dde4
To: <sip:sip.voicetrading.com>
Contact: <sip:Unknown at 91.205.172.196>
Call-ID: 2f0f8e621d8d84f52ce2f2006a07239d at 91.205.172.196
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 27 Jan 2010 13:55:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
#
U 2010/01/27 15:55:20.875367 194.120.0.198:5060 -> 91.205.172.196:5060
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 91.205.172.196:5060;branch=z9hG4bK15310eaf;rport
From: "Unknown" <sip:Unknown at 91.205.172.196>;tag=as3f83dde4
To: <sip:sip.voicetrading.com>
Contact: sip:194.120.0.198:5060
Call-ID: 2f0f8e621d8d84f52ce2f2006a07239d at 91.205.172.196
CSeq: 102 OPTIONS
Supported: foo
User-Agent: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Accept: application/sdp
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