[Asterisk-video] is this possible? asterisk -> sip voip provider -> gsm network -> 3g phone

Fri Jan 29 02:44:27 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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