[SIPForum-discussion] 183 Session Progress with SDP missing info

[Aniella Juverdeanu]

Hi,

Actually the receiver of 183 should allow the SDP without a= line for G.729, because payload 18 is a static value as indicated by RFC 3551, so well defined (it is not dynamic that needs the definition). It is not a must to declare it, maybe just for easy reading of the message, but should not drop the call because of it. If it does, it is a problem should be fixed as not being standard. I am fed –up with vendors that do not follow standards and surprised by the ones you don’t expect to. I found so many bugs on Cisco side either BTS, CM, CUBE, you name it.

RFC 3551                    RTP A/V Profile                    July 2003


               PT   encoding    media type  clock rate   channels
                    name                    (Hz)
               ___________________________________________________
               0    PCMU        A            8,000       1
               1    reserved    A
               2    reserved    A
               3    GSM         A            8,000       1
               4    G723        A            8,000       1
               5    DVI4        A            8,000       1
               6    DVI4        A           16,000       1
               7    LPC         A            8,000       1
               8    PCMA        A            8,000       1
               9    G722        A            8,000       1
               10   L16         A           44,100       2
               11   L16         A           44,100       1
               12   QCELP       A            8,000       1
               13   CN          A            8,000       1
               14   MPA         A           90,000       (see text)
               15   G728        A            8,000       1
               16   DVI4        A           11,025       1
               17   DVI4        A           22,050       1
               18   G729        A            8,000       1
               19   reserved    A
               20   unassigned  A
               21   unassigned  A
               22   unassigned  A
               23   unassigned  A
               dyn  G726-40     A            8,000       1
               dyn  G726-32     A            8,000       1
               dyn  G726-24     A            8,000       1
               dyn  G726-16     A            8,000       1
               dyn  G729D       A            8,000       1
               dyn  G729E       A            8,000       1
               dyn  GSM-EFR     A            8,000       1
               dyn  L8          A            var.        var.
               dyn  RED         A                        (see text)
               dyn  VDVI        A            var.        1

               Table 4: Payload types (PT) for audio encodings

From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Munjal Patel
Sent: June 11, 2012 3:36 PM
To: Lindström Tomi
Cc: discussion at sipforum.org
Subject: Re: [SIPForum-discussion] 183 Session Progress with SDP missing info

Tomi you were correct spoke with my carrier they found the issue and removed the ULC giving those bad 183's

Thanks everybody

Sent from my iPhone

On Jun 11, 2012, at 1:13 PM, Lindström Tomi <tomi.lindstrom at aina.fi<mailto:tomi.lindstrom at aina.fi>> wrote:
You are right there should be a-line for that g.729 and Cisco is working Like it supposed.

Br Tomi
________________________________
From: Munjal Patel
Sent: 11.06.2012 14:34
To: discussion at sipforum.org<mailto:discussion at sipforum.org>
Subject: [SIPForum-discussion] 183 Session Progress with SDP missing info
hi everyone i'm new to SIP and needing some help with a issue i'm having, any help would be appreciated

A= Cisco 2811 UAC
B= Carrier end point

During call setup carrier is responding back with 183 Session Progress SDP but with missing “a”-Line, here is the snips from the PCAP:


Invite- A to B (SIP/SDP)

SDP>
(M): audio 47640 RTP/AVP 18 101
(A): rtpmap:18 g729/8000/1
(A): rtpmap:101 telephone-event/8000/1
(A): ptime:20

100 Trying- B to A (SIP)

183 Session Progress- B to A (SIP/SDP)

SDP>
(M): audio 60050 RTP/AVP 18 101
(A): rtpmap:101 telephone-event/8000/1
(A): ptime:20

Cancel- A to B (SIP)


My question, is it normal for Cisco 2811 to send a cancel or can it be configured to try to renegotiate media with the carrier.


Thank You,

Munjal Patel
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