[Asterisk-Dev] FW: Payload numbers (X-Lite to Asterisk codec problem)

[Chris Heiser]

Rob,

Give this a shot:

Index: chan_sip.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.170
diff -u -r1.170 chan_sip.c
--- chan_sip.c  25 Aug 2003 14:17:14 -0000      1.170
+++ chan_sip.c  26 Aug 2003 14:32:50 -0000
@@ -119,7 +119,7 @@
 static int restart_monitor(void);

 /* Codecs that we support by default: */
-static int capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM
| AST_FORMAT_H263;
+static int capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM
| AST_FORMAT_H263 | AST_FORMAT_ILBC;
 static int noncodeccapability = AST_RTP_DTMF;

 static char ourhost[256];


--Chris

> -----Original Message-----
> From: asterisk-dev-admin at lists.digium.com [mailto:asterisk-dev-
> admin at lists.digium.com] On Behalf Of Rob
> Sent: Monday, August 25, 2003 11:12 PM
> To: asterisk-dev at lists.digium.com
> Subject: RE: [Asterisk-Dev] FW: Payload numbers (X-Lite to Asterisk codec
> problem)
> 
> 
> If a developer knowledgeable about the SDP/RTP payload code would like to
> test with a developer from XTen to fix the speex/iLBC problem, I'd be
> happy
> to set that up. Let me know.
> 
> -Rob
> 
> -----Original Message-----
> From: asterisk-dev-admin at lists.digium.com
> [mailto:asterisk-dev-admin at lists.digium.com]On Behalf Of Peter Grace
> Sent: August 22, 2003 4:58 PM
> To: asterisk-dev at lists.digium.com
> Subject: [Asterisk-Dev] FW: Payload numbers (X-Lite to Asterisk codec
> problem)
> 
> 
> 
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
> 
> Anyone know how to make sense of this?  I'm pretty dumb
> when it comes to the rtp protocol.  If someone can confirm
> this is a bug, we can put it into bugs.digium.com...
> 
> Pete
> 
> - -----Original Message-----
> From: Robin Raymond [mailto:robin at xten.com]
> Sent: Friday, August 22, 2003 6:48 PM
> To: Peter Grace
> Subject: RE: Payload numbers
> 
> 
> 
> Hi Peter,
> 
> Well, from what I understand by this user, Asterisk is not
> examining the RTP payload number for speex or iLBC that
> X-Lite is using.
> 
> In the INVITE request, I tell Asterisk which codec payload
> numbers I'm using, which is different that what Asterisk
> sends back. Asterisk reports back it's payload numbers. I
> send my RTP using your speex/iLBC payload numbers that you
> expect from your SDP, and I expect RTP to come back with my
> speex/iLBC payload numbers that I provided in my SDP. Now
> I'm not 100% sure I'm doing that right, so perhaps I have a
> bug.
> 
> v=0
> o=21186 21339250 21339250 IN IP4 64.180.128.147
> s=X-PRO
> c=IN IP4 64.180.128.147
> t=0 0
> m=audio 8000 RTP/AVP 0 8 3 98 97 101
> a=rtpmap:0 pcmu/8000
> a=rtpmap:8 pcma/8000
> a=rtpmap:3 gsm/8000
> a=rtpmap:98 iLBC/8000 <-- this is my mapping
> a=rtpmap:97 speex/8000 <-- this is my mapping
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> 
> As what I understand, calling w/ speex/iLBC simply doesn't
> work between X-Lite and Asterisk. Not sure why.
> 
> No I didn't receive the e-mail about the PPC. Feel free to
> CC me on stuff like that :)
> 
> Any ideas?
> 
> - -Rob
> 
> 
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> 
> 
> 
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