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

[Banda, Srinivas (Srinivas)]

Hi Kevin,

Codec 18 in the 183 SDP and INVITE  is treated as G729.
And Ptime in offer and answer are 20 msec. So ideally the call should
have connected. Not sure why CISCO switch is sending the CANCEL message?

Regards
Srinivas

-----Original Message-----
From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Kevin P. Fleming
Sent: Wednesday, June 13, 2012 8:10 PM
To: discussion at sipforum.org
Subject: Re: [SIPForum-discussion] 183 Session Progress with SDP missing
info

On 06/12/2012 03:33 AM, Andrey Ermoshin wrote:
> Hi, I think that one reason could be that there is no any valid codec
> offer in 183 message (18 101 payloads exists, but no g729 offer in
> SDP).

This is incorrect. The SDP is the 183 response below is well formed and 
RFC compliant. RTP payload number 18 is defined as being for G.729 at 
8kHz sample rate with one channel, so there is no requirement for an 
a:rtpmap line to appear in an SDP defining it.

>
> 2012/6/8 Munjal Patel<munjal84 at gmail.com>:
>> 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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Kevin P. Fleming
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