[Asterisk-video] 3G to SIP problem
Sergio García Murillo
Sergio.Garcia at ydilo.com
Thu Nov 16 06:34:26 MST 2006
Uff!!
Then you'll need a H324M gateway in the middle to handle the negotiation and decode both streams, at least
until someone finally develope it for asterisk at last.. :)
Greetings
Sergio
________________________________
From: asterisk-video-bounces at lists.digium.com [mailto:asterisk-video-bounces at lists.digium.com] On Behalf Of Nikolay Milovanov
Sent: jueves, 16 de noviembre de 2006 13:51
To: Development discussion of video media support in Asterisk
Subject: Re: [Asterisk-video] 3G to SIP problem
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 <mailto: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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>> >
>> >
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>
>
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--
Klaus Darilion
nic.at
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