[asterisk-users] g729a compatibility
Jeff LaCoursiere
jeff at jeff.net
Thu Jul 2 08:19:36 CDT 2009
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<jeff at jeff.net> 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<murdocke at gmail.com> 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<kpfleming at digium.com>
>>>> 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: kpfleming at digium.com
>>>>> Check us out at www.digium.com & www.asterisk.org
>>>>>
>>>>> _______________________________________________
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>>>>
>>>
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>
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