[Freeswitch-users] rtpmap question PCMU/PCMA

Steve Underwood steveu at coppice.org
Mon May 31 10:42:28 PDT 2010


Hi,

I think its a perfectly reasonable invite, including the shiny new 
capabilities stuff which should reach full RFC status shortly. As it is 
new, I think the jury is currently out on whether existing poorly 
implemented SIP packages will choke on it.

Steve


On 06/01/2010 12:35 AM, Michael Jerris wrote:
> This seems to be a badly broken sdp attempting to offer audio and t.38 but missing the m=image line from the sdp.
>
> Mike
>
> On May 31, 2010, at 10:57 AM, Mark Campbell-Smith wrote:
>
>    
>> Hi David,
>>
>> Its an INVITE.  Full invite below:
>>
>>    INVITE sip:gw+Phonzo at 124.xxx.xxx.xx:5080;transport=udp;gw=Phonzo SIP/2.0
>>    Record-Route:<sip:80.232.37.178;ftag=262787ae2a9104a0c7700794a69028aco;lr>
>>    Via: SIP/2.0/UDP
>> 80.232.37.178;branch=z9hG4bK8a48.ecc54191b911cf6bfd73daeefeae0ada.0
>>    Via: SIP/2.0/UDP
>> 80.232.37.178:5061;branch=z9hG4bK443b032c2a924807acf39718946e2c9e;rport=5061
>>    Max-Forwards: 16
>>    From: 010711xxxx
>> <sip:711xxxx at 80.232.37.178>;tag=262787ae2a9104a0c7700794a69028aco
>>    To:<sip:4610xxxxxxx at 80.232.37.178>
>>    Call-ID: M2Q1NjhhNmZjOGJjMDc3ODhlNzUyYzRiM2ZkMjQyZTE.
>>    CSeq: 200 INVITE
>>    Contact: Anonymous<sip:80.232.37.178:5061>
>>    Expires: 300
>>    User-Agent: Sippy
>>    cisco-GUID: 1658214937-1822691807-2631794736-96895450
>>    h323-conf-id: 1658214937-1822691807-2631794736-96895450
>>    Content-disposition: session
>>    Content-Length: 364
>>    Content-Type: application/sdp
>>
>>    v=0
>>    o=Sippy 141730476 0 IN IP4 80.232.37.178
>>    s=-
>>    t=0 0
>>    m=audio 47676 RTP/AVP 8 0 18 101
>>    c=IN IP4 213.50.91.3
>>    a=fmtp:18 annexb=yes
>>    a=rtpmap:101 telephone-event/8000
>>    a=fmtp:101 0-15
>>    a=sqn: 0
>>    a=cdsc: 1 audio RTP/AVP  8
>>    a=cdsc: 2 image udptl t38
>>    a=cpar: a=T38FaxUdpEC:t38UDPRedundancy
>>    a=cpar: a=T38FaxVersion:0
>>    a=cpar: a=T38MaxBitRate:14400
>>    a=sendrecv
>>
>> Regards
>> Mark
>>
>> On Mon, May 31, 2010 at 4:29 PM, David Ponzone<david.ponzone at gmail.com>  wrote:
>>      
>>> Mark,
>>> This looks like a T38 Re-INVITE, but a weird one.
>>> David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>> Ce message et toutes les pièces jointes sont confidentiels et établis à
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autorisée est interdite. Tout message électronique est susceptible
>>> d'altération. IPeva décline toute responsabilité au titre de ce message s'il
>>> a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce
>>> message, merci de le détruire immédiatement et d'avertir l'expéditeur.
>>>
>>>
>>>
>>> Le 31/05/2010 à 16:12, Mark Campbell-Smith a écrit :
>>>
>>> Hi All,
>>>
>>> I'm sure I've discussed this before, but I searched through my gmail
>>> and google and couldn't find the answer.
>>>
>>> Below is the SDP parameters from my sip provider.  Is it mandatory to
>>> always include the rtpmap details for PCMU/PCMA codes?  For example
>>> something like 'a=rtpmap:8 PCMA/8000' and 'a=rtpmap:0 PCMU/8000'?
>>>
>>> I'm using sipdroid on android and it rejects this with 'codec not supported'
>>>
>>> Thanks
>>>
>>>
>>>
>>> o=Sippy 141730476 0 IN IP4 xxx.xxx.xxx.xxx
>>> s=-
>>> t=0 0
>>> m=audio 47676 RTP/AVP 8 0 18 101
>>> c=IN IP4 xxx.xxx.xxx.xxx
>>> a=fmtp:18 annexb=yes
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-15
>>> a=sqn: 0
>>> a=cdsc: 1 audio RTP/AVP  8
>>> a=cdsc: 2 image udptl t38
>>> a=cpar: a=T38FaxUdpEC:t38UDPRedundancy
>>> a=cpar: a=T38FaxVersion:0
>>> a=cpar: a=T38MaxBitRate:14400
>>> ]
>>>        
>
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