[Freeswitch-users] rtpmap question PCMU/PCMA
Mark Campbell-Smith
mcampbellsmith at gmail.com
Wed Jun 2 07:14:36 PDT 2010
Coming back to my original question:
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'?
Is it possible to add the end of the SDP parameters by doing something like:
<extension name="fix_phonzo" continue="true">
<condition field="caller_id_number" expression="123456">
<action application="export"
data="sip_append_audio_sdp=a=rtpmap:8 PCMA/8000"/>
<action application="export"
data="sip_append_audio_sdp=a=rtpmap:0 PCMU/8000"/>
</condition>
</extension>
Can this be done anywhere in the dialplan as long as its before the
bridge command?
On Mon, May 31, 2010 at 9:15 PM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> 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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>>
>>
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