[Asterisk-video] Bridging 3g call with SIP

[m.ricordeau at newtech.fr]

Hello again,

I have done a tcpdump on outgoing sip on asterisk1
and found sip header for video is set with :
v=0
o=root 29048 29048 IN IP4 192.168.56.10
s=session
c=IN IP4 192.168.56.10
b=CT:384
t=0 0
m=audio 18514 RTP/AVP 96 0 8 101
a=rtpmap:96 AMR/8000
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 12692 RTP/AVP 34 103
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=sendrecv


but video packet are not sent.

I presume I need asterisk videocaps (oej team) to deal with video packets ?



Le Wed, 11 Feb 2009 19:23:22 +0100,
Michael Ricordeau <m.ricordeau at newtech.fr> a écrit :

> 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
> 
>