[cisco-voip] [OSL | CCIE_Voice] IPIPGW Sip to Sip

[Jonathan Charles]

CUCM is CCM 4.1.3(sr7)
Yes, the same behavior is exhibited in both directions.
G711ulaw all the way thru... if I change it to SIP to H323, it works fine in
both directions. No transcoders in the network.
Region is 711
The MTP required check box is set on the SIP trunk on CCM.



Jonathan

On Wed, Aug 6, 2008 at 2:45 PM, Chris Ward <chrward at cisco.com> wrote:

>  If the call works until the phone is answered, then it is probably
> media/capability related.
>
> First, is this the call flow?
>
> IP Phone --- CUCM --- SIP Trunk --- IPIPGW --- SIP --- CUCME --- IP Phone
>
> Now these questions:
>
>
>    1. What is the version of CUCM?
>    2. Do calls fail in both directions?
>    3. What is the region setting (what codec) between the CUCM IP Phone
>    and the SIP trunk on CUCM?
>    4. Is the CUCM SIP trunk doing early media (is the "MTP required"
>    checkbox checked)?
>
>
> -Chris
>
> ------------------------------
> *From: *Jonathan Charles <jonvoip at gmail.com>
> *Date: *Wed, 6 Aug 2008 14:32:31 -0500
> *To: *Stephen Collinson <scollinson at capewave.co.uk>
> *Cc: *OSL CCIE Voice Lab Exam <ccie_voice at onlinestudylist.com>, cisco voip
> <cisco-voip at puck.nether.net>
> *Subject: *Re: [cisco-voip] [OSL | CCIE_Voice] IPIPGW Sip to Sip
>
>
> Right, the question is, how do you configure it correctly?
>
> What would cuz the audio to not cut thru and the call to drop... I was
> suspecting codec, but it is G711 all the way thru (hard coded on each dial
> peer)
>
>
>
> Jonathan
>
> On Wed, Aug 6, 2008 at 2:21 PM, Stephen Collinson <
> scollinson at capewave.co.uk> wrote:
>
> SIP to SIP should work fine, when configured correctly.
>
>
>
> I was just trying to give you a scenario where we may need to use it.
> Apologies if this was not helpful
>
>
>
>
>
>
>   ------------------------------
>
> *From:* Jonathan Charles [mailto:jonvoip at gmail.com <jonvoip at gmail.com>]
> *Sent:* 06 August 2008 19:55
> *To:* Stephen Collinson
> *Cc:* cisco voip; OSL CCIE Voice Lab Exam
> *Subject:* Re: [OSL | CCIE_Voice] IPIPGW Sip to Sip
>
>
>
> Perhaps I wasn't clear...
>
>
> There is no CUE.
>
> This is a SCCP phone on a CCME, and a SCCP phone on CCM with a SIP trunk to
> an IPIPGW, and a SIP dial-peer to CCME...
>
>
> Jonathan
>
> On Wed, Aug 6, 2008 at 1:52 PM, Stephen Collinson <
> scollinson at capewave.co.uk> wrote:
>
> Perhaps worth looking at your config.
>
>
>
> You will need sip to sip, say to access CUE VM from a CCM SIP trunk.
>
>
>
> Check all G711 etc.
>
>
>
> Debug CCSIP
>
>
>
>
>
>
>   ------------------------------
>
> *From:* ccie_voice-bounces at onlinestudylist.com [
> mailto:ccie_voice-bounces at onlinestudylist.com<ccie_voice-bounces at onlinestudylist.com>]
> *On Behalf Of *Jonathan Charles
> *Sent:* 06 August 2008 18:41
>
>
> *To:* OSL CCIE Voice Lab Exam; cisco-voip at puck.nether.net
> *Subject:* [OSL | CCIE_Voice] IPIPGW Sip to Sip
>
>
>
> So, I was playing with an IPIPGW
>
>
>
> CCM on one side (SIP trunk) and CCME on the other (SIP dial-peer)... call
> worked, but as soon as you answered it dropped.
>
> I changed the SIP dial-peer from the IPIPGW to H.323 (no session protocol)
> and RTP cuts thru fine...
>
> Am I misreading something, is SIP to SIP not supported, or is my config
> retarded?
>
>
>
> Jonathan
>
>
>
>
> ------------------------------
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