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@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@80.232.37.178>;tag=262787ae2a9104a0c7700794a69028aco To:<sip:4610xxxxxxx@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@gmail.com> wrote:
Mark, This looks like a T38 Re-INVITE, but a weird one. David Ponzone Direction Technique email: david.ponzone@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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