[Asterisk-video] 3G to SIP transfer
Sergio Garcia Murillo
sergio.garcia at fontventa.com
Tue Dec 4 04:27:14 CST 2007
Have you applied also the patch for negotiation?
http://download.ives.fr/opensource/asterisk/videocodec_nego_fix_ast-1.4.13.patch.gz
BR
Sergio
----- Original Message -----
From: tech at i6net.com [mailto:tech at i6net.com]
To: asterisk-video at lists.digium.com
Sent: Tue, 04 Dec 2007 11:06:08 +0100
Subject: Re: [Asterisk-video] 3G to SIP transfer
Hi,
When I call from the SIP phone (ulaw), I can access to the video service (play
and record mp4 file with AMR stream).
So I suppose that the transcoder works with the video service.
I have "patched" the Asterisk with the AMR support from fontventa (I have check
the changes, and it seems to be OK).
Have someone succeed in transfering a 3G call to a Video SIP phone ?
Thanks,
Tech from i6net
Selon Mitul Limbani <mitul at enterux.com>:
> Hello,
>
> Do you have AMR codec in your SIP Phone ?
> I think that might be causing the problem.
>
> Thanks & Regards,
> Mitul Limbani,
> Founder & CEO,
> Enterux Solutions,
> The Enterprise Linux Company (TM),
> www.enterux.com
>
> Quoting tech at i6net.com:
>
> > Hello alls,
> >
> > I am trying to transfer a 3G video call to a SIP outgoing call using the
> Dial
> > command.
> > Asterisk seems to fail searching a codec translator (I don't know if it's
> for
> > the video or the audio stream).
> >
> > [Dec 3 17:37:57] WARNING[18752]: channel.c:3014 set_format: Unable to find
> a
> > codec translation path from unknown to unknown
> > [Dec 3 17:37:57] WARNING[18752]: channel.c:3395
> ast_channel_make_compatible:
> > Unable to set read format on channel Local/dial at default-4934,2 to 524288
> > [Dec 3 17:37:57] WARNING[18752]: app_dial.c:1640 dial_exec_full: Had to
> drop
> > call because I couldn't make Local/dial at default-4934,2 compatible with
> > SIP/octavius.i6net.org-0830c088
> >
> > Have someone an idea ?
> >
> > Thanks,
> >
> >
> > Tech from i6net
> >
> >
> > Here the full Asterisk CLI traces :
> >
> > quartus*CLI> sip debug
> > SIP Debugging re-enabled
> > -- Accepting call from '699435965' to '912104507' on channel 0/5, span 1
> > -- Executing [912104507 at default:1] Answer("Zap/5-1", "") in new stack
> > -- Executing [912104507 at default:2] h324m_gw("Zap/5-1", "dial at default")
> in
> > new stack
> > -- Executing [dial at default:1]
> h324m_gw_answer("Local/dial at default-4934,2",
> > "") in new stack
> > -- Executing [dial at default:2] Dial("Local/dial at default-4934,2",
> > "SIP/600 at octavius.i6net.org") in new stack
> > Video is at 193.22.119.85 port 10004
> > Audio is at 193.22.119.85 port 10050
> > Adding codec 0x2000 (amr) to SDP
> > Adding codec 0x8 (alaw) to SDP
> > Adding codec 0x4 (ulaw) to SDP
> > Adding codec 0x80000 (h263) to SDP
> > Adding codec 0x100000 (h263p) to SDP
> > Adding non-codec 0x1 (telephone-event) to SDP
> > Reliably Transmitting (no NAT) to 62.22.9.77:5060:
> > INVITE sip:600 at octavius.i6net.org SIP/2.0
> > Via: SIP/2.0/UDP 193.22.119.85:5060;branch=z9hG4bK052dfe35;rport
> > From: "699435965" <sip:699435965 at 193.22.119.85>;tag=as1d80f1af
> > To: <sip:600 at octavius.i6net.org>
> > Contact: <sip:699435965 at 193.22.119.85>
> > Call-ID: 2d9b830c0f58f7d72fe1ab4d4d708cdf at 193.22.119.85
> > CSeq: 102 INVITE
> > User-Agent: Divina
> > Max-Forwards: 70
> > Date: Mon, 03 Dec 2007 16:37:57 GMT
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> > Supported: replaces
> > Content-Type: application/sdp
> > Content-Length: 397
> >
> > v=0
> > o=root 17239 17239 IN IP4 193.22.119.85
> > s=session
> > c=IN IP4 193.22.119.85
> > b=CT:384
> > t=0 0
> > m=audio 10050 RTP/AVP 96 8 0 101
> > a=rtpmap:96 AMR/8000
> > a=rtpmap:8 PCMA/8000
> > a=rtpmap:0 PCMU/8000
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-16
> > a=silenceSupp:off - - - -
> > a=ptime:20
> > a=sendrecv
> > m=video 10004 RTP/AVP 34 103
> > a=rtpmap:34 H263/90000
> > a=rtpmap:103 h263-1998/90000
> > a=sendrecv
> >
> > ---
> > -- Called 600 at octavius.i6net.org
> > [Dec 3 17:37:57] WARNING[18752]: channel.c:3014 set_format: Unable to find
> a
> > codec translation path from unknown to unknown
> > [Dec 3 17:37:57] WARNING[18752]: channel.c:3395
> ast_channel_make_compatible:
> > Unable to set read format on channel SIP/octavius.i6net.org-0830c088
> > to 524288
> > quartus*CLI>
> > <--- SIP read from 62.22.9.77:5060 --->
> > SIP/2.0 100 Trying
> > Via: SIP/2.0/UDP
> > 193.22.119.85:5060;branch=z9hG4bK052dfe35;received=193.22.119.85;rport=5060
> > From: "699435965" <sip:699435965 at 193.22.119.85>;tag=as1d80f1af
> > To: <sip:600 at octavius.i6net.org>
> > Call-ID: 2d9b830c0f58f7d72fe1ab4d4d708cdf at 193.22.119.85
> > CSeq: 102 INVITE
> > User-Agent: Divina
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> > Supported: replaces
> > Contact: <sip:600 at 62.22.9.77>
> > Content-Length: 0
> >
> >
> > <------------->
> > --- (11 headers 0 lines) ---
> > quartus*CLI>
> > <--- SIP read from 62.22.9.77:5060 --->
> > SIP/2.0 200 OK
> > Via: SIP/2.0/UDP
> > 193.22.119.85:5060;branch=z9hG4bK052dfe35;received=193.22.119.85;rport=5060
> > From: "699435965" <sip:699435965 at 193.22.119.85>;tag=as1d80f1af
> > To: <sip:600 at octavius.i6net.org>;tag=as7133a043
> > Call-ID: 2d9b830c0f58f7d72fe1ab4d4d708cdf at 193.22.119.85
> > CSeq: 102 INVITE
> > User-Agent: Divina
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> > Supported: replaces
> > Contact: <sip:600 at 62.22.9.77>
> > Content-Type: application/sdp
> > ontent-Length: 366
> >
> > v=0
> > o=root 19599 19599 IN IP4 62.22.9.77
> > s=session
> > c=IN IP4 62.22.9.77
> > b=CT:384
> > t=0 0
> > m=audio 10024 RTP/AVP 8 0 101
> > a=rtpmap:8 PCMA/8000
> > a=rtpmap:0 PCMU/8000
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-16
> > a=silenceSupp:off - - - -
> > a=ptime:20
> > a=sendrecv
> > m=video 10026 RTP/AVP 34 103
> > a=rtpmap:34 H263/90000
> > a=rtpmap:103 h263-1998/90000
> > a=sendrecv
> >
> > <------------->
> > --- (12 headers 18 lines) ---
> > Found RTP audio format 8
> > Found RTP audio format 0
> > Found RTP audio format 101
> > Found RTP video format 34
> > Found RTP video format 103
> > Peer audio RTP is at port 62.22.9.77:10024
> > Found description format PCMA for ID 8
> > Found description format PCMU for ID 0
> > Found description format telephone-event for ID 101
> > Found description format H263 for ID 34
> > Found description format h263-1998 for ID 103
> > Capabilities: us - 0x18000c (ulaw|alaw|h263|h263p), peer - audio=0x18000c
> > (ulaw|alaw|h263|h263p)/video=0x180000 (h263|h263p), combined - 0x18000c
> > (ulaw|alaw|h263|h263p)
> > Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
> > (telephone-event), combined - 0x1 (telephone-event)
> > Peer audio RTP is at port 62.22.9.77:10024
> > Peer video RTP is at port 62.22.9.77:10026
> > list_route: hop: <sip:600 at 62.22.9.77>
> > set_destination: Parsing <sip:600 at 62.22.9.77> for address/port to send to
> > set_destination: set destination to 62.22.9.77, port 5060
> > Transmitting (no NAT) to 62.22.9.77:5060:
> > ACK sip:600 at 62.22.9.77 SIP/2.0
> > Via: SIP/2.0/UDP 193.22.119.85:5060;branch=z9hG4bK39bd2334;rport
> > From: "699435965" <sip:699435965 at 193.22.119.85>;tag=as1d80f1af
> > To: <sip:600 at octavius.i6net.org>;tag=as7133a043
> > Contact: <sip:699435965 at 193.22.119.85>
> > Call-ID: 2d9b830c0f58f7d72fe1ab4d4d708cdf at 193.22.119.85
> > CSeq: 102 ACK
> > User-Agent: Divina
> > Max-Forwards: 70
> > Content-Length: 0
> >
> >
> > ---
> > -- SIP/octavius.i6net.org-0830c088 answered Local/dial at default-4934,2
> > [Dec 3 17:37:57] WARNING[18752]: channel.c:3014 set_format: Unable to find
> a
> > codec translation path from unknown to unknown
> > [Dec 3 17:37:57] WARNING[18752]: channel.c:3395
> ast_channel_make_compatible:
> > Unable to set read format on channel Local/dial at default-4934,2 to 524288
> > [Dec 3 17:37:57] WARNING[18752]: app_dial.c:1640 dial_exec_full: Had to
> drop
> > call because I couldn't make Local/dial at default-4934,2 compatible with
> > SIP/octavius.i6net.org-0830c088
> > Scheduling destruction of SIP dialog
> > '2d9b830c0f58f7d72fe1ab4d4d708cdf at 193.22.119.85' in 32000 ms (Method:
> INVITE)
> > set_destination: Parsing <sip:600 at 62.22.9.77> for address/port to send to
> > set_destination: set destination to 62.22.9.77, port 5060
> > Reliably Transmitting (no NAT) to 62.22.9.77:5060:
> > BYE sip:600 at 62.22.9.77 SIP/2.0
> > Via: SIP/2.0/UDP 193.22.119.85:5060;branch=z9hG4bK23d8c153;rport
> > From: "699435965" <sip:699435965 at 193.22.119.85>;tag=as1d80f1af
> > To: <sip:600 at octavius.i6net.org>;tag=as7133a043
> > Call-ID: 2d9b830c0f58f7d72fe1ab4d4d708cdf at 193.22.119.85
> > CSeq: 103 BYE
> > User-Agent: Divina
> > Max-Forwards: 70
> > Content-Length: 0
> >
> >
> > ---
> > == Spawn extension (default, dial, 2) exited non-zero on
> > 'Local/dial at default-4934,2'
> > == Spawn extension (default, 912104507, 2) exited non-zero on 'Zap/5-1'
> > -- Hungup 'Zap/5-1'
> > quartus*CLI>
> > <--- SIP read from 62.22.9.77:5060 --->
> > SIP/2.0 200 OK
> > Via: SIP/2.0/UDP
> > 193.22.119.85:5060;branch=z9hG4bK23d8c153;received=193.22.119.85;rport=5060
> > From: "699435965" <sip:699435965 at 193.22.119.85>;tag=as1d80f1af
> > To: <sip:600 at octavius.i6net.org>;tag=as7133a043
> > Call-ID: 2d9b830c0f58f7d72fe1ab4d4d708cdf at 193.22.119.85
> > CSeq: 103 BYE
> > User-Agent: Divina
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> > Supported: replaces
> > Contact: <sip:600 at 62.22.9.77>
> > Content-Length: 0
> >
> >
> > <------------->
> > --- (11 headers 0 lines) ---
> > Really destroying SIP dialog
> '2d9b830c0f58f7d72fe1ab4d4d708cdf at 193.22.119.85'
> > Method: INVITE
> >
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>
>
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