[Asterisk-video] 3G to SIP problem

[Nikolay Milovanov]

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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