[Asterisk-video] 3G to SIP problem

Thu Nov 16 06:55:45 MST 2006

Hopefully, I'll release soon :)

Subject: RE: [Asterisk-video] 3G to SIP problemDate: Thu, 16 Nov 2006 14:34:26 +0100From: Sergio.Garcia at ydilo.comTo: asterisk-video at lists.digium.com

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 MilovanovSent: jueves, 16 de noviembre de 2006 13:51To: Development discussion of video media support in AsteriskSubject: 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 DarilionSent: jueves, 16 de noviembre de 2006 12:02To: Development discussion of video media support in Asterisk Subject: Re: [Asterisk-video] 3G to SIP problemHi!I guess these lines are irrelevant, because the m line offers only codec 125.regardsklausNikolay 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>> >>> >>> > _______________________________________________ >> > --Bandwidth and Colocation provided by Easynews.com -->> >>> > asterisk-video mailing list>> > To UNSUBSCRIBE or update options visit: >> >>> > http://lists.digium.com/mailman/listinfo/asterisk-video>> >>> >>> >>> _______________________________________________ >> --Bandwidth and Colocation provided by Easynews.com -->>>> asterisk-video mailing list>> To UNSUBSCRIBE or update options visit:>>     http://lists.digium.com/mailman/listinfo/asterisk-video>>>>> ---------------------------------------------------------------------- > -->> _______________________________________________> --Bandwidth and Colocation provided by Easynews.com -->> asterisk-video mailing list> To UNSUBSCRIBE or update options visit: >    http://lists.digium.com/mailman/listinfo/asterisk-video--Klaus Darilionnic.at_______________________________________________ --Bandwidth and Colocation provided by Easynews.com --asterisk-video mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-video--------------------------------------------------------------------------------------This message and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. 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