[SIPForum-discussion] What is the difference between 408 and 504 responses
seshagiri
seshgirik at gmail.com
Tue Jun 26 03:36:35 UTC 2012
Sorry i posted in existing thread , re-posting my querty , Pl give your
comments:
On Mon, Jun 25, 2012 at 11:17 AM, seshagiri <seshgirik at gmail.com> wrote:
> What is the difference between 408 and 504 responses ?
>
>
> In the below negative scenarios our sbc forward the recieived requests
> to UAS.
> UAS will not respond for few requests( It will not send 200 for INFO or
> UPDATE or OPTIONS)
> In some scenarios UAS will not send 200 OK after 180.
> But our sbc sends responses like 408 or 504 to UAC.
> I could not able to understand when it sends 408 and when it sends 504 ?
>
>
>
>
> Messages Retrans Timeout Unexpected-Msg
> INVITE ----------> 0 0 0
> 100 <---------- 0 0 0 0
> 180 <---------- 0 0 0 0
> 183 <---------- 0 0 0 0
> 200 <---------- E-RTD1 0 0 0 0
> ACK ----------> 0 0
> Pause [ 1000ms] 0 0
> INFO ----------> 0 0
> 100 <---------- 0 0 0 0
> 408 <---------- E-RTD1 0 0 0 0
> Pause [ 20.0s] 0 0
> BYE ----------> 0 0 0
> 200 <---------- 0 0 0 0
>
>
> Messages Retrans Timeout
> Unexpected-Msg
> INVITE ----------> 0 0 0
> 100 <---------- 0 0 0 0
> 180 <---------- 0 0 0 0
>
> 504 <---------- E-RTD1 0 0 0 0
> ACK ----------> 0 0
>
>
>
> Messages Retrans Timeout
> Unexpected-Msg
> INVITE ----------> 0 0 0
> 100 <---------- 0 0 0 0
> 180 <---------- 0 0 0 0
> 183 <---------- 0 0 0 0
>
> PRACK ----------> 0 0
> 200 <---------- 0 0 0 0
> 504 <---------- 0 0 0 0
>
> ACK ----------> 0 0
>
>
> Messages Retrans Timeout
> Unexpected-Msg
> INVITE ----------> 1 0 0
> 100 <---------- 1 0 0 0
> 180 <---------- 0 0 0 0
> 183 <---------- 1 0 0 0
>
> PRACK ----------> 1 0
> 200 <---------- 1 0 0 0
> UPDATE ----------> 1 0
> 504 <---------- 0 0 0 0
>
> ACK ----------> 0 0
> ------- Waiting for active calls to end. Press [q] again to force exit.
> -------
>
>
>
> Messages Retrans Timeout
> Unexpected-Msg
> INVITE ----------> 1 0 0
> 100 <---------- 1 0 0 0
> 180 <---------- 1 0 0 0
> 183 <---------- 0 0 0 0
> 200 <---------- E-RTD1 1 0 0 0
> ACK ----------> 1 0
> Pause [ 1000ms] 1 0
> INFO ----------> 1 0
> 100 <---------- 0 0 0 0
> 408 <---------- E-RTD1 1 0 0 0
> Pause [ 20.0s] 1 0
> BYE ----------> 1 0 0
> 200 <---------- 1 0 0 0
>
> ------------------------------ Test Terminated
> --------------------------------
>
>
>
>
>
>
> Messages Retrans Timeout
> Unexpected-Msg^M
> INVITE ----------> 1 0 0 ^M
> 100 <---------- 1 0 0 0 ^M
> 180 <---------- 1 0 0 0 ^M
> 183 <---------- 0 0 0 0 ^M
> 200 <---------- E-RTD1 1 0 0 0 ^M
> ACK ----------> 1 0 ^M
> Pause [ 3000ms] 1 0 ^M
> REFER ----------> 1 5 0 ^M
> 100 <---------- 0 0 0 0 ^M
> 408 <---------- 1 0 0 0 ^M
> BYE ----------> 1 0 0 ^M
> 200 <---------- 1 0 0 0 ^M
> ^M
> ------------------------------ Test Terminated ---------------------
>
>
> Messages Retrans Timeout
> Unexpected-Msg^M
> ----------> INVITE 1 0 0 0 ^M
> ^M
> <---------- 180 1 0 ^M
> <---------- 200 1 0 ^M
> ----------> ACK E-RTD1 1 0 0 0 ^M
> ^M
> [ 3000ms] Pause 1 0 ^M
> <---------- OPTIONS 1 7 0 ^M
> ----------> 408 1 0 0 0 ^M
> ^M
> ----------> BYE 1 0 0 0 ^M
> <---------- 200 1 0 ^M
> ------------------------------ Test Terminated
> --------------------------------^M
> ^M
>
>
> Messages Retrans Timeout
> Unexpected-Msg^M
> NOTIFY ----------> 1 0 ^M
> 100 <---------- 0 0 0 0 ^M
> 408 <---------- E-RTD1 1 0 0 0 ^M
> ------------------------------ Test Terminated ----------------------------
>
>
> Regards
> Sesh
>
>
>
> On Thu, Jun 21, 2012 at 10:01 PM, Aniella Juverdeanu <
> Aniella.Juverdeanu at telus.com> wrote:
>
>> Allow header just lets know the remote party that INFO is supported as a
>> method but if the two SIP end points do not support out of band DTMF tones
>> in INFO message, then should use in-band as RFC 2833 RTP packets.****
>>
>> ** **
>>
>> *From:* Phillip Lewis [mailto:phillipl at joinvip.com]
>> *Sent:* June 18, 2012 6:40 PM
>> *To:* Aniella Juverdeanu
>> *Cc:* virajith; discussion
>> *Subject:* Re: [SIPForum-discussion] dtmf issue on the sip trunk !!****
>>
>> ** **
>>
>> What about the Allow field in the SIP Header. INFO may have been
>> included, could SIP INFO have been resolved.?****
>>
>> On Mon, Jun 18, 2012 at 11:45 AM, Aniella Juverdeanu <
>> Aniella.Juverdeanu at telus.com> wrote:****
>>
>> Hi,****
>>
>> ****
>>
>> To me it looks negotiated ok with CODEC G711and RFC 2833/4733 for DTMF
>> events 0-15. ****
>>
>> I didn’t see until now 77 and 84 used but if not accepted by the other
>> end it means not supported – but the call should continue with standard
>> events 0-15.****
>>
>> ****
>>
>> More details you may find in RFC 2833/4733.****
>>
>> ****
>>
>> Aniella ****
>>
>> ****
>>
>> *From:* discussion-bounces at sipforum.org [mailto:
>> discussion-bounces at sipforum.org] *On Behalf Of *virajith
>> *Sent:* June 12, 2012 9:47 PM
>> *To:* discussion
>> *Subject:* [SIPForum-discussion] dtmf issue on the sip trunk !!****
>>
>> ****
>>
>>
>> Hello Team,
>>
>> I have an issue where dtmf is not getting negotiated . We are doing a
>> delayed offer to pbx but the pbx ..
>>
>> What the dtmf methods advertised in this 200 ok sdp sent by the pbx ?
>>
>> v=0
>> o=IPC 425146 425146 IN IP4 XXXXX
>> s=IPC
>> c=IN IP4 10.97.232.239
>> t=0 0
>> m=audio 16442 RTP/AVP 0 8 18 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:18 G729/8000
>> a=fmtp:18 annexb=yes
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15,77,84
>> a=ptime:20
>> a=sendrecv
>>
>>
>> in the ACK that we(our end) send is this...
>>
>>
>> v=0
>> o=CiscoSystemsCCM-SIP 2000 1 IN IP4 10.97.202.101
>> s=SIP Call
>> c=IN IP4 10.97.202.124
>> t=0 0
>> m=audio 19504 RTP/AVP 0 101
>> a=rtpmap:0 PCMU/8000
>> a=ptime:20
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>>
>>
>> What the dtmf methods advertised and negotiated?
>>
>>
>> Thanks,
>> Vir
>>
>>
>>
>>
>>
>> ****
>>
>> [image: Image removed by sender.]<http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle?>
>> ****
>>
>> Follow *Rediff Deal ho jaye!<http://track.rediff.com/click?url=___http://dealhojaye.rediff.com?sc_cid=rediffmailsignature___&cmp=signature&lnk=rediffmailsignature&newservice=deals>
>> * to get exciting offers in your city everyday.****
>>
>> ****
>>
>>
>> _______________________________________________
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>> Post to the list at discussion at sipforum.org****
>>
>>
>>
>> ****
>>
>> ** **
>>
>> --
>> *Phillip Lewis*
>> Senior Manager, Engineering
>> *VIP Communications Inc*
>> Office | 703-708-1515 Ext 213
>> Fax | 703-708-1518
>> Email | phillipl at joinvip.com ****
>>
>> Website | www.joinvip.com****
>>
>> ** **
>>
>> _______________________________________________
>> This is the SIP Forum discussion mailing list
>> TO UNSUBSCRIBE, or edit your delivery options, please visit
>> http://sipforum.org/mailman/listinfo/discussion
>> Post to the list at discussion at sipforum.org
>>
>>
>
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