[Asterisk-video] 3G to SIP problem

Thu Nov 16 03:29:56 MST 2006

Hi,

a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 192-194,200-202

This is a probably a proprietary (non-standard) SDP content generated
by your 3G phone. I remember I read from some RFC (dont remember which
one already), that prefix X- is normally used to indicate that the
following is proprietary and understood only by the device, which
generated it. The rest of the devices not aware of this extension,
will simply ignore it.

  Best of luck,

   Andrey.

On 11/16/06, Nikolay Milovanov <n.milovanov at gmail.com> 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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