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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