[Asterisk-video] Bridging 3g call with SIP

Fri Feb 13 04:38:23 CST 2009

Hi,

I have make tests yesterday, so it's time for feedback ;)

On asterisk1 and asterisk2, I have tested : 
SIP -> 3GPhone
3GPhone -> SIP
and all is ok (this is a good news !)

Today, I have tested : 3Gphone to 3Gphone passing throw two asterisk :
 3Gphone --Zap--> [asterisk1] --Sip--> [asterisk2] --Zap--> 3Gphone
and it's working too .

The next try was : 3Gphone to 3G platform passing throw two asterisk :
 3Gphone --Zap--> [asterisk1] --Sip--> [asterisk2] --Zap--> 3Gplatform
and it's not working :
  call is up (in 3g mode) and on 3G platform, I can see
connected call is amr/h263 as expected.
  But in 3Gphone I didn't have audio and video.

So, 3G platform may have something wrong :
 bad data format ?

Is there a way to get informations asterisk2 is receiving from
3gplatform during a 3g call ? 
(Using h324m debug ?)

Note that 3G platform is Dylogic PSE 3G VAS Genie .

Thanks  

Le Thu, 12 Feb 2009 09:45:59 +0100,
Sergio Garcia Murillo <sergio.garcia at fontventa.com> a écrit :

> Hi Michael,
> 
> It should work without videocaps branch.
> 
> Why don't you test:
> 
> 3Gphone --Zap--> [asterisk1] --> Sip Phone
> 
> and 
> 
> Sip Phone--> [asterisk2] --Zap -- > 3Gphone
> 
> independently to check which part is failing you?
> 
> Best regards
> Sergio
> 
> Michael Ricordeau escribió:
> > Hello,
> >
> > is this possible to bridge with two asterisk :
> >
> > 3Gphone --Zap--> [asterisk1] --Sip--> [asterisk2] --Zap-->
> > 3Gplatform
> >
> >
> >
> > I'm calling with 3G mobile phone and try to get 3Gplatform stream
> > back to phone.
> >
> >
> >
> > In asterisk1 and asterisk2,
> > codec amr, h263, h263p are ok :
> >         INT    BINARY        HEX   TYPE       NAME   DESC
> > --------------------------------------------------------------------------------
> >           1 (1 <<  0)      (0x1)  audio       g723   (G.723.1)
> >           2 (1 <<  1)      (0x2)  audio        gsm   (GSM)
> >           4 (1 <<  2)      (0x4)  audio       ulaw   (G.711 u-law)
> >           8 (1 <<  3)      (0x8)  audio       alaw   (G.711 A-law)
> >          16 (1 <<  4)     (0x10)  audio   g726aal2   (G.726 AAL2)
> >          32 (1 <<  5)     (0x20)  audio      adpcm   (ADPCM)
> >          64 (1 <<  6)     (0x40)  audio       slin   (16 bit Signed
> > Linear PCM) 128 (1 <<  7)     (0x80)  audio      lpc10   (LPC10)
> >         256 (1 <<  8)    (0x100)  audio       g729   (G.729A)
> >         512 (1 <<  9)    (0x200)  audio      speex   (SpeeX)
> >        1024 (1 << 10)    (0x400)  audio       ilbc   (iLBC)
> >        2048 (1 << 11)    (0x800)  audio       g726   (G.726 RFC3551)
> >        4096 (1 << 12)   (0x1000)  audio       g722   (G722)
> >        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)
> >
> >
> > Applications H324M (and lib) are ok on both servers.
> >
> > I have set on both sip.conf :
> > videosupport=yes
> > disallow=all
> > allow=h263p
> > allow=h263
> > allow=h264
> > allow=h261
> > allow=amr
> > allow=alaw
> > allow=ulaw
> >
> > Asterisk loopback test Video_loopback() is working on both servers.
> >
> >
> >
> > On asterisk1, extensions.conf is :
> > [default]
> > exten => 1483,1,H324m_gw(CALL at 3gp_videos)
> >
> > [3gp_videos]
> > exten => CALL,1,H324m_gw_answer()
> > exten => CALL,n,Dial(SIP/534444444 at 10.0.0.249)
> >
> >
> > On asterisk2, Sip is incoming on context from-sip with dialplan :
> > [from-sip]
> > exten => _53XXXXXXX,1,h324m_call(99${EXTEN}@from-sip)
> > exten => _9953XXXXXXX,1,Set(CHANNEL(transfercapability)=VIDEO)
> > exten =>
> > _9953XXXXXXX,n,NoOp(transfer=${CHANNEL(transfercapability)}) exten
> > => _9953XXXXXXX,n,Set(CHANNEL(userinformationlayer1)=38) exten =>
> > _9953XXXXXXX,n,NoOp(ul1=${CHANNEL(userinformationlayer1)}) exten =>
> > _9953XXXXXXX,n,Dial(Zap/r3/(${EXTEN:2}|20)
> >
> >
> >
> > I can see on asterisk2 :
> >
> >     -- digital call, setting user information layer 1 to 38 (0x26)
> >     -- Requested transfer capability: 0x18 - VIDEO
> >     -- Called r3/(534444444|20
> >     -- Zap/70-1 is proceeding passing it to
> > Local/99534320659 at from-sip-5f24,2 -- Zap/70-1 is ringing
> >     -- Channel 0/8, span 3 got hangup, cause 16
> >     -- Hungup 'Zap/70-1'
> >     -- No one is available to answer at this time (1:0/0/0)
> >   == Auto fallthrough, channel 'Local/99534320659 at from-sip-5f24,2'
> > status is 'NOANSWER' == Auto fallthrough, channel
> > 'SIP/192.168.56.10-08282c10' status is 'UNKNOWN'
> >
> >
> > If I tried another dialplan on asterisk2 with mp4play (just playing
> > a 3gp file), I only have audio .
> >
> > So, I don't know if I can bridge like that 3G calls . (I'm probably
> > on a wrong way ...)
> >
> >
> > Best Regards
> >
> >
> >   
> 
> 
> 
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-- 
Michaël Ricordeau
Email: m.ricordeau at newtech.fr
Tel: +33561434871 
Newtech Multimedia
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