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

Peter Grace pgrace at fierymoon.com
Tue Aug 26 17:16:22 MST 2003


 
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Chris,

	I gave this a shot (adding AST_FORMAT_ILBC) on my system
and calling between a 7960 and X-lite works fine,
appropriate recoding path found and all...  However,
VoiceMail2 (and possibly voicemail?) don't produce
intelligible sound when you connect to them.  I'm thinking
that AST_FORMAT_ILBC is half the problem, now all we have
to do is figure out why voicemail (and indeed, all IVR
prompts) are unintelligible.

	I also tried the same magic with SPEEX (adding
AST_FORMAT_SPEEX) and that didn't work at all -- no calls
could find appropriate decoding paths to each other.

Can we add those formats into chan_sip.c?  As long as the
user is calling another user, it seems as if ilbc works
like a charm.


Pete



- -----Original Message-----
From: asterisk-dev-admin at lists.digium.com
[mailto:asterisk-dev-admin at lists.digium.com] On Behalf Of
Chris Heiser
Sent: Tuesday, August 26, 2003 10:09 AM
To: asterisk-dev at lists.digium.com
Subject: RE: [Asterisk-Dev] FW: Payload numbers (X-Lite to
Asterisk codec problem)


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)
> 
> 
> 
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> 
> 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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