[asterisk-users] 0005162: RTP Packetization : Few questions

[yusuf]

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:01 at 192.168.0.64 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.196:5060;branch=z9hG4bK6fcf8559;rport
From: "asterisk" <sip:asterisk at 192.168.0.196>;tag=as5a8f594f
To: <sip:01 at 192.168.0.64>
Contact: <sip:asterisk at 192.168.0.196>
Call-ID: 21534acc1dfc61fd64133d850f73b11b at 192.168.0.196
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 - 21534acc1dfc61fd64133d850f73b11b at 192.168.0.196
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:asterisk at 192.168.0.196>
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:asterisk at 192.168.0.196>;tag=as5a8f594f
To: <sip:01 at 192.168.0.64>
Call-ID: 21534acc1dfc61fd64133d850f73b11b at 192.168.0.196
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:01 at 192.168.0.64>
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:asterisk at 192.168.0.196>;tag=as5a8f594f
To: <sip:01 at 192.168.0.64>;tag=as63837eba
Call-ID: 21534acc1dfc61fd64133d850f73b11b at 192.168.0.196
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:01 at 192.168.0.64>
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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