[Asterisk-video] Bridging 3g call with SIP

[Michael Ricordeau]

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


-- 
Michaël Ricordeau