[Asterisk-video] 3G to SIP problem

Mon Nov 20 03:31:57 MST 2006

Yes. Klaus. In this case the Asterisk or the IP PBX should do that.
Unfortunately it is still not so advanced :(  Hopefully in the future.

On 11/20/06, Klaus Darilion <klaus.mailinglists at pernau.at> wrote:
>
> Yes. looks like the gateway does not demultiplex the video and audio but
> just forwards the raw ISDN B channel.
>
> regards
> klaus
>
> Nikolay Milovanov wrote:
> > Hi Sergio,
> >
> > I guess that's because of the clear channel. For me that means that both
> > are
> > encoded in it.
> >
> > Thanks for the help.
> >
> > Niko
> >
> >
> > On 11/16/06, Sergio García Murillo <Sergio.Garcia at ydilo.com> wrote:
> >>
> >> Appart of that, no video media is specified in the sdp.
> >>
> >> Greetings
> >>         Sergio
> >>
> >> -----Original Message-----
> >> From: asterisk-video-bounces at lists.digium.com [mailto:
> >> asterisk-video-bounces at lists.digium.com] On Behalf Of Klaus Darilion
> >> Sent: jueves, 16 de noviembre de 2006 12:02
> >> To: Development discussion of video media support in Asterisk
> >> Subject: Re: [Asterisk-video] 3G to SIP problem
> >>
> >> Hi!
> >>
> >> I guess these lines are irrelevant, because the m line offers only
> codec
> >> 125.
> >>
> >> regards
> >> klaus
> >>
> >> Nikolay Milovanov wrote:
> >> > Thanks Andrey,
> >> >
> >> > the call is actually a video call and the video is comming from the
> >> > softswitch as unrestricted digital G.nX64/8000). I guess asterisk in
> >> > general is not supporting unrestricted digital.
> >> >
> >> > Could somebody explain to me what is that means:
> >> >
> >> > a=X-cpar: a=rtpmap:100 X-NSE/8000
> >> > a=X-cpar: a=fmtp:100 192-194,200-202
> >> >
> >> >
> >> > BR,
> >> > Niko
> >> >
> >> > On 11/16/06, Andrey Kuprianov < andrey.kouprianov at gmail.com> wrote:
> >> >>
> >> >> Yup,
> >> >>
> >> >> It looks like Asterisk does not support your codec. That's what your
> >> >> SDP
> >> >> says:
> >> >>
> >> >> a=rtpmap:125 G.nX64/8000
> >> >> a=rtpmap:101 /8000
> >> >> a=rtpmap:100 /8000
> >> >>
> >> >> And that's what you have in config file:
> >> >>
> >> >> allow=alaw
> >> >> allow=speex
> >> >> allow=gsm
> >> >>
> >> >> Try switching codec to one of these listed in your sip.conf.
> >> >>
> >> >>
> >> >> On 11/16/06, Nikolay Milovanov <n_milovanov at mail.bg> wrote:
> >> >> > Hi Guys,
> >> >> >
> >> >> > My Scenario is
> >> >> >
> >> >> > 3Gphone -> (3G network provider)->(Softswitch Cisco
> >> >> > PGW)->SIP<-Asterisk<-SIP->SIP phone
> >> >> >
> >> >> > I am using Asterisk 1.4 beta 3.  I am calling from 3G to SIP. As I
> >> >> > see
> >> >> from
> >> >> > the trace Asterisk is not supporting the clear chanel codec
> >> >> (a=rtpmap:125
> >> >> > G.nX64/8000) used by the PGW.
> >> >> >
> >> >> > Am I right or the problem is somewhere else? Please take a look of
> >> >> > my
> >> >> config
> >> >> > and the trace of the asterisk cli.
> >> >> >
> >> >> >
> >> >> > sip.conf
> >> >> >
> >> >> > [general]
> >> >> >
> >> >> > videosupport=yes
> >> >> > disallow=all                    ; First disallow all codecs
> >> >> > allow=alaw                      ; Allow codecs in order of
> >> preference
> >> >> > allow=h263
> >> >> > allow=h263p
> >> >> > allow=h261
> >> >> >
> >> >> > [32515901]
> >> >> > type=friend
> >> >> > secret=phone1
> >> >> > host=dynamic
> >> >> > ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
> >> >> > mailbox=1000 ; Mailbox for message waiting indicator context=sip
> >> >> > videosupport=yes
> >> >> > maxcallbitrate=128
> >> >> > callerid= "test" <32515901>
> >> >> > allow=alaw
> >> >> > allow=speex
> >> >> > allow=gsm
> >> >> > allow=h261
> >> >> > allow=h263
> >> >> > allow=h263p
> >> >> >
> >> >> >
> >> >> > Debug
> >> >> >
> >> >> > <--- SIP read from my.domain.com:5060 ---> INVITE
> >> >> > sip:32515901 at 172.18.10.100;user=phone SIP/2.0
> >> >> > Via: SIP/2.0/UDP my.domain.com:5060
> >> >> > ;branch=z9hG4bKterm-30-myphone-32515901-17145
> >> >> > From: myphone <sip:myphone at my.domain.com
> ;user=phone>;tag=1763495500
> >> >> > To: 32515901 < sip:32515901 at 172.18.10.100 ;user=phone>
> >> >> > Call-ID: 3f7a5361-3057ba40-5f7fc171-25 at my.domain.com
> >> >> > CSeq: 1 INVITE
> >> >> > Supported: timer
> >> >> > Session-Expires: 1800
> >> >> > Min-SE: 1800
> >> >> > Contact:  <sip:myphone at my.domain.com:5060>
> >> >> > Allow:
> >> >> > INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE
> >> >> > Max-Forwards: 70
> >> >> > Content-Type: application/sdp
> >> >> > Content-Length: 317
> >> >> >
> >> >> > v=0
> >> >> > c=IN IP4 85.118.195.7
> >> >> > m=audio 18010 RTP/AVP 125
> >> >> > a=rtpmap:125 G.nX64/8000
> >> >> > a=X-pc-codec: 125 101 100
> >> >> > a=rtpmap:125 G.nX64/8000
> >> >> > a=rtpmap:101 /8000
> >> >> > a=rtpmap:100 /8000
> >> >> > a=X-sqn:0
> >> >> > a=X-cap: 1 audio RTP/AVP 100
> >> >> > a=X-cpar: a=rtpmap:100 X-NSE/8000
> >> >> > a=X-cpar: a=fmtp:100 192-194,200-202
> >> >> > a=X-cap: 2 image udptl t38
> >> >> >
> >> >> > <------------->
> >> >> > --- (14 headers 13 lines) ---
> >> >> > Using INVITE request as basis request -
> >> >> > 3f7a5361-3057ba40-5f7fc171-25 at my.domain.com
> >> >> > Found peer 'test'
> >> >> > Found RTP audio format 125
> >> >> > Peer audio RTP is at port 85.118.195.7:18010 Found description
> >> >> > format G.nX64 for ID 125 Found description format G.nX64 for ID
> 125
> >> >> > Capabilities: us - 0x1c0008 (alaw|h261|h263|h263p), peer -
> >> >> > audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
> >> >> > Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer -
> >> >> > 0x0 (nothing), combined - 0x0 (nothing) [Nov 15 23:15:53]
> >> >> > NOTICE[22446]: chan_sip.c:4996 process_sdp: No
> >> >> compatible
> >> >> > codecs, not accepting this offer!
> >> >> >
> >> >> > <--- Reliably Transmitting (no NAT) to my.domain.com:5060 --->
> >> >> > SIP/2.0 488 Not acceptable here
> >> >> > Via: SIP/2.0/UDP my.domain.com:5060
> >> >> > ;branch=z9hG4bKterm-30-myphone-32515901-17145;received=
> >> >> > my.domain.com
> >> >> > From: myphone <
> >> >> > sip:myphone at my.domain.com;user=phone>;tag=1763495500
> >> >> > To: 32515901 <
> >> >> > sip:32515901 at 172.18.10.100;user=phone>;tag=as11b984c5
> >> >> > Call-ID: 3f7a5361-3057ba40-5f7fc171-25 at my.domain.com
> >> >> > CSeq: 1 INVITE
> >> >> > User-Agent: Asterisk PBX
> >> >> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> >> >> > Supported: replaces
> >> >> > Content-Length: 0
> >> >> >
> >> >> > Appreciate any help,
> >> >> >
> >> >> > Niko
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > --Bandwidth and Colocation provided by Easynews.com --
> >> >> >
> >> >> > asterisk-video mailing list
> >> >> > To UNSUBSCRIBE or update options visit:
> >> >> >
> >> >> > http://lists.digium.com/mailman/listinfo/asterisk-video
> >> >> >
> >> >> >
> >> >> >
> >> >> _______________________________________________
> >> >> --Bandwidth and Colocation provided by Easynews.com --
> >> >>
> >> >> asterisk-video mailing list
> >> >> To UNSUBSCRIBE or update options visit:
> >> >>    http://lists.digium.com/mailman/listinfo/asterisk-video
> >> >>
> >> >
> >> >
> >> >
> ----------------------------------------------------------------------
> >> > --
> >> >
> >> > _______________________________________________
> >> > --Bandwidth and Colocation provided by Easynews.com --
> >> >
> >> > asterisk-video mailing list
> >> > To UNSUBSCRIBE or update options visit:
> >> >    http://lists.digium.com/mailman/listinfo/asterisk-video
> >>
> >>
> >> --
> >> Klaus Darilion
> >> nic.at
> >>
> >> _______________________________________________
> >> --Bandwidth and Colocation provided by Easynews.com --
> >>
> >> asterisk-video mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>    http://lists.digium.com/mailman/listinfo/asterisk-video
> >>
> >>
> --------------------------------------------------------------------------------------
> >>
> >> This message and any files transmitted with it are confidential and
> >> intended solely
> >> for the use of the individual or entity to whom they are addressed. No
> >> confidentiality
> >> or privilege is waived or lost by any wrong transmission.
> >> If you have received this message in error, please immediately destroy
> it
> >> and kindly
> >> notify the sender by reply email.
> >> You must not, directly or indirectly, use, disclose, distribute,
> >> print, or
> >> copy any
> >> part of this message if you are not the intended recipient. Opinions,
> >> conclusions and
> >> other information in this message that do not relate to the official
> >> business of
> >> Ydilo Advanced Voice Solutions, S.A. shall be understood as neither
> given
> >> nor endorsed by it.
> >>
> >>
> --------------------------------------------------------------------------------------
> >>
> >> _______________________________________________
> >> --Bandwidth and Colocation provided by Easynews.com --
> >>
> >> asterisk-video mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>    http://lists.digium.com/mailman/listinfo/asterisk-video
> >>
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-video mailing list
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-video
>
>
> --
> Klaus Darilion
> nic.at
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-video
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-video/attachments/20061120/c782e2dc/attachment-0001.htm