[Asterisk-video] Bridging 3g call with SIP

Wed Feb 11 17:42:27 CST 2009

Hi Michael,

- Have you try just with enabling the h263p codec ?
- Do you have patched the Asterisk to enable 3G video outgoing calls ?
(Small question, are you trying to find a solution for the expensive 3G 
video call roaming ?)

Regards,

Borja

m.ricordeau at newtech.fr a écrit :
> 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
>>
>>
>>     
>
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