Hi Sergio, <br><br>I guess that's because of the clear channel. For me that means that both are encoded in it. <br><br>Thanks for the help. <br><br>Niko<br><br><br><div><span class="gmail_quote">On 11/16/06, <b class="gmail_sendername">
Sergio García Murillo</b> <<a href="mailto:Sergio.Garcia@ydilo.com">Sergio.Garcia@ydilo.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Appart of that, no video media is specified in the sdp.<br><br>Greetings<br> Sergio<br><br>-----Original Message-----<br>From: <a href="mailto:asterisk-video-bounces@lists.digium.com">asterisk-video-bounces@lists.digium.com
</a> [mailto:<a href="mailto:asterisk-video-bounces@lists.digium.com">asterisk-video-bounces@lists.digium.com</a>] On Behalf Of Klaus Darilion<br>Sent: jueves, 16 de noviembre de 2006 12:02<br>To: Development discussion of video media support in Asterisk
<br>Subject: Re: [Asterisk-video] 3G to SIP problem<br><br>Hi!<br><br>I guess these lines are irrelevant, because the m line offers only codec 125.<br><br>regards<br>klaus<br><br>Nikolay Milovanov wrote:<br>> Thanks Andrey,
<br>><br>> the call is actually a video call and the video is comming from the<br>> softswitch as unrestricted digital G.nX64/8000). I guess asterisk in<br>> general is not supporting unrestricted digital.<br>
><br>> Could somebody explain to me what is that means:<br>><br>> a=X-cpar: a=rtpmap:100 X-NSE/8000<br>> a=X-cpar: a=fmtp:100 192-194,200-202<br>><br>><br>> BR,<br>> Niko<br>><br>> On 11/16/06, Andrey Kuprianov <
<a href="mailto:andrey.kouprianov@gmail.com">andrey.kouprianov@gmail.com</a>> wrote:<br>>><br>>> Yup,<br>>><br>>> It looks like Asterisk does not support your codec. That's what your<br>>> SDP
<br>>> says:<br>>><br>>> a=rtpmap:125 G.nX64/8000<br>>> a=rtpmap:101 /8000<br>>> a=rtpmap:100 /8000<br>>><br>>> And that's what you have in config file:<br>>><br>>> allow=alaw
<br>>> allow=speex<br>>> allow=gsm<br>>><br>>> Try switching codec to one of these listed in your sip.conf.<br>>><br>>><br>>> On 11/16/06, Nikolay Milovanov <<a href="mailto:n_milovanov@mail.bg">
n_milovanov@mail.bg</a>> wrote:<br>>> > Hi Guys,<br>>> ><br>>> > My Scenario is<br>>> ><br>>> > 3Gphone -> (3G network provider)->(Softswitch Cisco<br>>> > PGW)->SIP<-Asterisk<-SIP->SIP phone
<br>>> ><br>>> > I am using Asterisk 1.4 beta 3. I am calling from 3G to SIP. As I<br>>> > see<br>>> from<br>>> > the trace Asterisk is not supporting the clear chanel codec<br>>> (a=rtpmap:125
<br>>> > G.nX64/8000) used by the PGW.<br>>> ><br>>> > Am I right or the problem is somewhere else? Please take a look of<br>>> > my<br>>> config<br>>> > and the trace of the asterisk cli.
<br>>> ><br>>> ><br>>> > sip.conf<br>>> ><br>>> > [general]<br>>> ><br>>> > videosupport=yes<br>>> > disallow=all ; First disallow all codecs
<br>>> > allow=alaw ; Allow codecs in order of preference<br>>> > allow=h263<br>>> > allow=h263p<br>>> > allow=h261<br>>> ><br>>> > [32515901]<br>
>> > type=friend<br>>> > secret=phone1<br>>> > host=dynamic<br>>> > ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info<br>>> > mailbox=1000 ; Mailbox for message waiting indicator context=sip
<br>>> > videosupport=yes<br>>> > maxcallbitrate=128<br>>> > callerid= "test" <32515901><br>>> > allow=alaw<br>>> > allow=speex<br>>> > allow=gsm<br>>> > allow=h261
<br>>> > allow=h263<br>>> > allow=h263p<br>>> ><br>>> ><br>>> > Debug<br>>> ><br>>> > <--- SIP read from <a href="http://my.domain.com:5060">my.domain.com:5060
</a> ---> INVITE<br>>> > <a href="mailto:sip:32515901@172.18.10.100">sip:32515901@172.18.10.100</a>;user=phone SIP/2.0<br>>> > Via: SIP/2.0/UDP <a href="http://my.domain.com:5060">my.domain.com:5060</a>
<br>>> > ;branch=z9hG4bKterm-30-myphone-32515901-17145<br>>> > From: myphone <<a href="mailto:sip:myphone@my.domain.com">sip:myphone@my.domain.com</a>;user=phone>;tag=1763495500<br>>> > To: 32515901 <
<a href="mailto:sip:32515901@172.18.10.100">sip:32515901@172.18.10.100</a> ;user=phone><br>>> > Call-ID: <a href="mailto:3f7a5361-3057ba40-5f7fc171-25@my.domain.com">3f7a5361-3057ba40-5f7fc171-25@my.domain.com
</a><br>>> > CSeq: 1 INVITE<br>>> > Supported: timer<br>>> > Session-Expires: 1800<br>>> > Min-SE: 1800<br>>> > Contact: <<a href="http://sip:myphone@my.domain.com:5060">sip:myphone@my.domain.com:5060
</a>><br>>> > Allow:<br>>> > INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE<br>>> > Max-Forwards: 70<br>>> > Content-Type: application/sdp<br>>> > Content-Length: 317
<br>>> ><br>>> > v=0<br>>> > c=IN IP4 <a href="http://85.118.195.7">85.118.195.7</a><br>>> > m=audio 18010 RTP/AVP 125<br>>> > a=rtpmap:125 G.nX64/8000<br>>> > a=X-pc-codec: 125 101 100
<br>>> > a=rtpmap:125 G.nX64/8000<br>>> > a=rtpmap:101 /8000<br>>> > a=rtpmap:100 /8000<br>>> > a=X-sqn:0<br>>> > a=X-cap: 1 audio RTP/AVP 100<br>>> > a=X-cpar: a=rtpmap:100 X-NSE/8000
<br>>> > a=X-cpar: a=fmtp:100 192-194,200-202<br>>> > a=X-cap: 2 image udptl t38<br>>> ><br>>> > <-------------><br>>> > --- (14 headers 13 lines) ---<br>>> > Using INVITE request as basis request -
<br>>> > <a href="mailto:3f7a5361-3057ba40-5f7fc171-25@my.domain.com">3f7a5361-3057ba40-5f7fc171-25@my.domain.com</a><br>>> > Found peer 'test'<br>>> > Found RTP audio format 125<br>>> > Peer audio RTP is at port
<a href="http://85.118.195.7:18010">85.118.195.7:18010</a> Found description<br>>> > format G.nX64 for ID 125 Found description format G.nX64 for ID 125<br>>> > Capabilities: us - 0x1c0008 (alaw|h261|h263|h263p), peer -
<br>>> > audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)<br>>> > Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer -<br>>> > 0x0 (nothing), combined - 0x0 (nothing) [Nov 15 23:15:53]
<br>>> > NOTICE[22446]: chan_sip.c:4996 process_sdp: No<br>>> compatible<br>>> > codecs, not accepting this offer!<br>>> ><br>>> > <--- Reliably Transmitting (no NAT) to <a href="http://my.domain.com:5060">
my.domain.com:5060</a> ---><br>>> > SIP/2.0 488 Not acceptable here<br>>> > Via: SIP/2.0/UDP <a href="http://my.domain.com:5060">my.domain.com:5060</a><br>>> > ;branch=z9hG4bKterm-30-myphone-32515901-17145;received=
<br>>> > <a href="http://my.domain.com">my.domain.com</a><br>>> > From: myphone <<br>>> > <a href="mailto:sip:myphone@my.domain.com">sip:myphone@my.domain.com</a>;user=phone>;tag=1763495500
<br>>> > To: 32515901 <<br>>> > <a href="mailto:sip:32515901@172.18.10.100">sip:32515901@172.18.10.100</a>;user=phone>;tag=as11b984c5<br>>> > Call-ID: <a href="mailto:3f7a5361-3057ba40-5f7fc171-25@my.domain.com">
3f7a5361-3057ba40-5f7fc171-25@my.domain.com</a><br>>> > CSeq: 1 INVITE<br>>> > User-Agent: Asterisk PBX<br>>> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>>> > Supported: replaces
<br>>> > Content-Length: 0<br>>> ><br>>> > Appreciate any help,<br>>> ><br>>> > Niko<br>>> ><br>>> ><br>>> > _______________________________________________
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