[Asterisk-video] 3G to SIP problem
Klaus Darilion
klaus.mailinglists at pernau.at
Mon Nov 20 02:51:28 MST 2006
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
>> >> >
>> >> >
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>> --
>> Klaus Darilion
>> nic.at
>>
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Klaus Darilion
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