Re: TR: I-D Action:draft-boucadair-sipping-ipv6-atypes-00.txt

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

 




________________________________________
> From: Henry Sinnreich [mailto:hsinnrei@xxxxxxxxx]
> Sent: Thursday, March 05, 2009 9:25 AM
> To: Hadriel Kaplan; Dan Wing; mohamed.boucadair@xxxxxxxxxxxxxxxxxx; sipping@xxxxxxxx
> Subject: Re:  TR: I-D Action:draft-boucadair-sipping-ipv6-atypes-00.txt
>
> Could you venture to give an intuitive estimate of the % of cases where ICE does not work?
> In the scenarios described here.

Nope, I have no idea what the % is, because we don't track that separately.  It's a lot more than one would think though, because essentially SIP peering and SIP-Trunks work that way.  Your call started from your house or Enterprise, into a Service Provider, and from there if the call gets routed to a peering partner, the media relay path setup would go through the access SBC's between you and the provider, and then through the peering SBC's into the next provider, and so on. Or the SIP call started on a Mobile Switch, CO switch, or PSTN gateway, and if it gets routed out of the PoP or domain to another provider or for example for international termination, the media gets relayed through the peering SBCs to specific peering trunks.

Or if you make an E911 call, the PSAP distributor has its own SBC's, which relay the media for the specific calls that get routed to it, but sometimes release the media when they then get routed to specific PSAPs.  And of course if your call is lawfully intercepted, it gets relayed.  And I know that being able to bridge calls between Enterprise VPN's is a popular feature, because we used to win deals over competitors who couldn't do that.  And I know of operators that have MPLS-TE LSPs between SBC's for voice traffic, to do shortcut routing and/or fast-reroute protection.

And a lot of providers' PSTN gateways for inbound/outbound calls and voicemail systems are in a private network, and they don't do ICE, so the only way to get your media into that private network is by relaying it, but if the call ends up going to another SIP subscriber it may get released.

-hadriel
_______________________________________________
Sipping mailing list  https://www.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@xxxxxxxxxxxxxxx for questions on current sip
Use sip@xxxxxxxx for new developments of core SIP

[Index of Archives]     [IETF Announce]     [IETF Discussion]     [Linux SCSI]     [Linux USB Devel]     [Video for Linux]     [Linux Audio Users]     [Yosemite News]     [Linux Kernel]     [Linux SCSI]     [XFree86]     [Big List of Linux Books]

  Powered by Linux