[Asterisk-video] Bridging 3g call with SIP
Michael Ricordeau
m.ricordeau at newtech.fr
Thu Feb 12 03:41:12 CST 2009
Hi Borja,
thanks for your reply.
1 - Have you try just with enabling the h263p codec ?
Just trying with codecs restricted to h263p and amr in sip.conf on
both asterisk servers and I have the same problem :
On asterisk2 :
-- Executing [534320659 at from-sip:1] Set("SIP/192.168.56.10-08265118", "CHANNEL(transfercapability)=DIGITAL") in new stack
-- Executing [534320659 at from-sip:2] NoOp("SIP/192.168.56.10-08265118", "transfer=DIGITAL") in new stack
-- Executing [534320659 at from-sip:3] Set("SIP/192.168.56.10-08265118", "CHANNEL(userinformationlayer1)=38") in new stack
-- Executing [534320659 at from-sip:4] NoOp("SIP/192.168.56.10-08265118", "ul1=38") in new stack
-- Executing [534320659 at from-sip:5] Dial("SIP/192.168.56.10-08265118", "Zap/r3/(534444444|20") in new stack
-- digital call, setting user information layer 1 to 38 (0x26)
-- Requested transfer capability: 0x08 - DIGITAL
-- Called r3/(534444444|20
-- Zap/72-1 is proceeding passing it to SIP/192.168.56.10-08265118
-- Zap/72-1 is ringing
-- Channel 0/10, span 3 got hangup, cause 16
-- Hungup 'Zap/72-1'
-- No one is available to answer at this time (1:0/0/0)
== Auto fallthrough, channel 'SIP/192.168.56.10-08265118' status is
'NOANSWER'
2 - Do you have patched the Asterisk to enable 3G video outgoing calls ?
This path ?
http://download.ives.fr/opensource/asterisk/videocodec_nego_fix_ast-1.4.13.patch.gz
So I didn't patch asterisk with this one.
Only installed asterisk from this howto :
http://asterisk-party.org/index.php/Asterisk_Video_3G_EN
with asterisk 1.4.21
And for the small question, it's not for 3G roaming but this is the same approach :
I have ~ 10 E1 connected to 3 asterisk : A1, A2, A3.
I have a 3g platform (accessible throw Asterisk Zap) connected to A1.
Currently only 3g call incoming from E1 connected to A1 can go throw 3g platform (3g passthrough
work with no problem with userinformationlayer1 and transfercapability
=> patches for proper h324m ZAP signaling by klaus) .
I want 3g call from E1 connected to A2 and A3 can be forwarded to A1 throw SIP protocol and A1
bridge incoming SIP to Zap/3G platform .
Best regards
Le Thu, 12 Feb 2009 00:42:27 +0100,
Borja SIXTO <borja.sixto at i6net.com> a écrit :
> 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
> >>
> >>
> >>
> >
> > _______________________________________________
> > --Bandwidth and Colocation Provided by http://www.api-digital.com--
> >
> > asterisk-video mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-video
> >
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-video
--
Michaël Ricordeau
Email: m.ricordeau at newtech.fr
Tel: +33561434871
Newtech Multimedia
3, chemin du Pigeonnier de la cépière
BP 53575
31035 Toulouse Cedex 1
More information about the asterisk-video
mailing list