[Freeswitch-users] rtpmap question PCMU/PCMA

Mark Campbell-Smith mcampbellsmith at gmail.com
Mon May 31 12:15:35 PDT 2010


Thanks Guys.

I should have noted that this is not related to an FS fault at all.
When using FS in bypass media mode, the call is rejected by the
client.  Without bypass media mode it works.

I just know there is a huge sip knowledge on this mailing list, and
would be able to get my answers easily.

I guess the android client sipdroid does not like the broken and/or
shiny new SDP format.

Cheers

On Mon, May 31, 2010 at 7:42 PM, Steve Underwood <steveu at coppice.org> wrote:
> 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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