[Asterisk-video] 3G to SIP problem

Thu Nov 16 05:51:04 MST 2006

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
> >> >
> >> >
> >> > _______________________________________________
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>
> --
> Klaus Darilion
> nic.at
>
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