[Asterisk-video] 3G to SIP problem

Thu Nov 16 04:12:35 MST 2006

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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> 
> 
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--
Klaus Darilion
nic.at

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