Dan Austin wrote:
As far as the above is concerned I have the following:
I am using Asterisk 1.2.10, patched with this patch for 1.2.10.
I have 2 * boxes. They call each other over SIP, and I have in
sip.conf on both boxes
autoframing=yes
disallow=all
allow=g729:80
When A calls B, it sets ptime:80.
On B I see this:
We're at 192.168.0.64 port 11004
Adding codec 0x100 (g729) to SDP
Sep 7 18:16:16 WARNING[5529]: frame.c:1072 ast_codec_pref_getsize:
Framing not set for codec g729, using default 20 and ptime:20
I'll have a look at the 1.2.10 patch
So B is setting packetization to 20, when it should be 80, and is not
respecting autoframing.
Another developer wrote the autoframing feature, and I have not used
it, but I'll look to see if there is an obvious reason why it does
not find or honor the ptime.
Can you capture the SIP INVITE dialog on box B so I can see the SDP
offer, and look to see if the ptime element is present and set
properly?
Here is the capture: (here packetization is set to 60)
196 is A, initiated the call
64 is B, recieved the call
<-- SIP read from 192.168.0.196:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.196:5060;branch=z9hG4bK6fcf8559;rport
From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as5a8f594f
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 07 Sep 2006 16:17:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 251
v=0
o=root 5447 5447 IN IP4 192.168.0.196
s=session
c=IN IP4 192.168.0.196
t=0 0
m=audio 16146 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:60
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
--- (13 headers 12 lines)---
Using INVITE request as basis request - [EMAIL PROTECTED]
Sending to 192.168.0.196 : 5060 (NAT)
Found no matching peer or user for '192.168.0.196:5060'
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.196:16146
Found description format G729
Found description format telephone-event
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing),
combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1
(telephone-event)
Looking for 01 in default (domain 192.168.0.64)
list_route: hop: <sip:[EMAIL PROTECTED]>
Transmitting (NAT) to 192.168.0.196:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.0.196:5060;branch=z9hG4bK6fcf8559;received=192.168.0.196;rport=5060
From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as5a8f594f
To: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
---
-- Executing NoOp("SIP/192.168.0.196-09fea900", "YUSUF") in new stack
-- Executing Playback("SIP/192.168.0.196-09fea900", "demo-congrats") in new
stack
We're at 192.168.0.64 port 11004
Adding codec 0x100 (g729) to SDP
Sep 7 18:16:16 WARNING[5529]: frame.c:1072 ast_codec_pref_getsize: Framing not set for codec g729,
using default 20
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.0.196:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.196:5060;branch=z9hG4bK6fcf8559;received=192.168.0.196;rport=5060
From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as5a8f594f
To: <sip:[EMAIL PROTECTED]>;tag=as63837eba
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[EMAIL PROTECTED]>
Content-Type: application/sdp
Content-Length: 249
v=0
o=root 5494 5494 IN IP4 192.168.0.64
s=session
c=IN IP4 192.168.0.64
t=0 0
m=audio 11004 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
--
thanks,
yusuf
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