On Thu, 2 Jul 2009, Elliot Murdock wrote:
Hello Jeff,
Yes, I use G729 all the time.
Here is the SDP extrace from Wireshark. I'll get more data as it
becomes available:
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): MG4000|2.0 49743 83164 IN
IP4 216.48.184.27
Owner Username: MG4000|2.0
Session ID: 49743
Session Version: 83164
Owner Network Type: IN
Owner Address Type: IP4
Owner Address: 216.48.184.27
Session Name (s): -
Connection Information (c): IN IP4 216.48.184.27
Connection Network Type: IN
Connection Address Type: IP4
Connection Address: 216.48.184.27
Time Description, active time (t): 0 0
Session Start Time: 0
Session Stop Time: 0
Media Description, name and address (m): audio 25184
RTP/AVP 18 98 96 97 101 13
Media Type: audio
Media Port: 25184
Media Proto: RTP/AVP
Media Format: ITU-T G.729
Media Format: 98
Media Format: 96
Media Format: 97
Media Format: 101
Media Format: Comfort noise
Media Attribute (a): rtpmap:98 G.729a/8000
Media Attribute Fieldname: rtpmap
Media Format: 98
MIME Type: G.729a
Media Attribute (a): rtpmap:96 G.729ab/8000
Media Attribute Fieldname: rtpmap
Media Format: 96
MIME Type: G.729ab
Media Attribute (a): rtpmap:97 G.729b/8000
Media Attribute Fieldname: rtpmap
Media Format: 96
MIME Type: G.729ab
Media Attribute (a): rtpmap:97 G.729b/8000
Media Attribute Fieldname: rtpmap
Media Format: 97
MIME Type: G.729b
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute Fieldname: rtpmap
Media Format: 101
MIME Type: telephone-event
Media Attribute (a): fmtp:101 0-15
Media Attribute Fieldname: fmtp
Media Format: 101 [telephone-event]
Media format specific parameters: 0-15
Media Attribute (a): fmtp:18 annexb=no
Media Attribute Fieldname: fmtp
Media Format: 18 [telephone-event]
Media format specific parameters: annexb=no
Media Attribute (a): ptime:20
Media Attribute Fieldname: ptime
Media Attribute Value: 20
Media Attribute (a): rtpmap:13 CN/8000
Media Attribute Fieldname: rtpmap
Media Format: 13
MIME Type: CN
Thank you,
Elliot
Please stop top posting - it is making it impossible to follow the thread.
This is the offer from your device (what is it?). Where is the reply?
Please post the relevant section of your sip.conf.
j
On Thu, Jul 2, 2009 at 4:04 PM, Jeff LaCoursiere<[email protected]> wrote:
On Thu, 2 Jul 2009, Elliot Murdock wrote:
Hello!
Which RFC specifies the corresponding number of the formats?
Where in the Asterisk source code does it state the SDP formats?
Does Asterisk follow the formats of IANA?
(http://www.iana.org/assignments/rtp-parameters)
Thank you,
Elliot
Perhaps this is falling back too far, but do you have G.729 licenses for
your asterisk server?
j
On Thu, Jul 2, 2009 at 3:44 PM, Elliot Murdock<[email protected]> wrote:
Hello,
Thank you clarifying that.
However, if that is the case, why is Asterisk sending back PCMU
packets (instead of G729), which the device is not enabled for and
subsequently, fails the call?
Could the mapping be disabled or not properly mapping to the G729
driver in a certain versions of Asterisk?
Thanks,
Elliot
On Thu, Jul 2, 2009 at 3:25 PM, Kevin P. Fleming<[email protected]>
wrote:
Elliot Murdock wrote:
Hello!
I noticed that the SIP packet contains this line:
m=audio 60000 RTP/AVP 18 98 96 97 101 13
However, there is no rtpmap that describes 18. Media format 18
Apparently refers to G729, but there is no rtpmap in the SDP for it.
Since G729 is a registered and known format is there any way for
Asterisk to negotiate it within an explicit rtpmap?
Yes, that is already supported. Asterisk does not require rtpmap entries
for well-known (RFC specified) codec mappings.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: [email protected]
Check us out at www.digium.com & www.asterisk.org
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users