一些和sip协议相关的FAQ(5)

Is there a SIP interoperability certification? How can I test interoperability with others?

There currently is no certification that attests to the functionality and compatibility of a SIP implementation. However, there are regular SIP bake-offs where implementors can test their work. Also, some sites have set up public SIP servers.

mailto:islepchin@dynamicsoft.com?subject=SIP FAQ

2000-Jul-04 11:27pm

Why use SIP-T as opposed to tunneling SS7 using SCTP?

Using SCTP (aka SIGTRAN) to send SS7 between softswitches works fine assuming that you know the terminating device is an SS7 enabled softswitch, and that you are not interested in services provided by SIP. By using SIP instead, a softswitch can talk the same call control protocol to other softswitches, PC clients, gateways, IP phones, and so on. Furthermore, the softswitch does not need to know the identity of the terminating device ahead of time. In real networks, it will be unlikely that the originating softswitch knows. Calls will terminate in networks owned by other providers, in which case the type of terminating device cannot be known ahead of time. SIP-T is ideal in that the extra ISUP information carried is ignored by non-SS7 devices, so it works for all devices.

mailto:jdrosen@dynamicsoft.com?subject=SIP FAQ

2000-Jul-05 11:25pm

How does SIP carry DTMF (touch tones)?

First, in most cases it is not clear that SIP is the right mechanism for this, since DTMF detection is being done in devices that generate RTP, not SIP.

RTP can be used to carry DTMF, as described in RFC 2833. RFC 2833 uses "forward error correction", retransmitting DTMF digits periodically. Thus, unless there are extremely long bursts of packet errors, digits are transmitted reliably. Retransmission by SIP, either at the application layer or via TCP, is based on exponential back-off, with delays of a few seconds after several consecutive losses. If a human generates the touch tone commands, it is possible that such long retransmission delays will cause the user to press the button again, resulting in duplicate digits.

DTMF over RTP is also required to synchronize audio and touch tones at VoIP-to-PSTN gateways.

Gateways that are only interested in detecting tones do not need to buffer audio and can simply forward the audio packets while doing playout buffering and DTMF detection locally.

A number of proposals exist for carrying DTMF in SIP INFO messages, but the working group has not decided which of the approaches, if any, to pursue.

2000-Jul-10 4:11pm

What is the relationship of a "softswitch" to SIP?

The term "softswitch" is primarily a marketing term, with no well-defined technical meaning. It is often used to designate a collection of software providing telephony interworking services, including a signaling gateway (SG), media gateway controller (MGC) and a SIP user agent (UA). It can use any number of protocols, depending on the particular application and network configuration, including ISUP, CAS, MGCP, Megaco, H.323 and SIP. Many "softswitches" use SIP for communicating between softswitches.

2000-Jul-20 12:06pm

How does SIP/SDP relate to T.38 fax calls?

An SDP addition to allow SIP/SDP to set up T.38 fax calls is specified in Annex D of ITU Rec. T.38 ("SIP/SDP Call Establishment Procedures"). It can also be found at ftp://standards.nortelnetworks.com/itu_to_ietf/SG8/February00/

2000-Jul-24 6:12pm

Where is the use of SIP defined in 3GPP?

The exact signalling and call control protocols are defined in 3GPP Technical Specification 3G TS 24.228: "Signalling flows for the IP multimedia call control based on SIP and SDP" and 3GPP Technical Specification 3G TS 24.229: "IP Multimedia Call Control Protocol based on SIP and SDP". (Jack Yu) 3GPP documents can be found at http://www.3gpp.org/

The draft version for TS24.228 is available at http://www.3gpp.org/ftp/Specs/Latest_drafts/ The draft version for TS24.229 is available at http://www.3gpp.org/ftp/Specs/Latest_drafts/

Also the current version for TS23.228 is available at http://www.3gpp.org/ftp/Specs/2001-12/Rel-5/23_series/

http://www.3gpp.org/TB/Other/IETF.htm lists the IETF documents that 3GPP depends on, including many SIP documents.

2002-Jan-08 10:17am

What port numbers should I use for RTP sessions set up with SIP?

You will need an even/odd port pair. The even port is for RTP, and the odd is for RTCP. Never use ports below 1024. It is strongly recommended that you avoid ports between 1024 and 49151, since these are assigned by IANA, and might be in use by some other protocol. It is highly recommended that you choose ports between 49152 and 65534. The general algorithm for this is to choose a pair in this range, and request it. If one or both are in use, try another pair until you obtain a free one.

mailto:jdrosen@dynamicsoft.com?subject=SIP FAQ

2001-Apr-22 9:47pm

Why use SIP for presence and instant messaging?

SIP networks already provide presence information (via REGISTER) as it is needed for session intitiation;

delivery of presence and notification messages to people is supported on top of existing proxy networks;

instant messages are often best modeled as a session;

presence and IM are both parts of a broader communications service that also includes voice, video, shared applications, etc., and it would be nice to smoothly integtrate those together,

events are needed for a number of session-related functions, thus SIP needs event notification functionality in any event (events are a generalization of presence);

a requirements analysis showed that much of what's needed for a scalable presence and IM protoco is already offered by SIP.

2001-Jul-26 3:30pm

How do I show this FAQ as a single page?

Click on [Show This Entire Category] at the bottom of the SIP FAQ "front" page, in the gray bar.

2000-Jul-10 4:03pm

RTP Issues

RTP FAQs are covered at http://www.cs.columbia.edu/~hgs/rtp.

In particular, a common mistake is to assume that RTP mandates a certain packetization interval, e.g., 20 ms. This is *wrong*. While RFC 1890 recommends certain values and SDP allows to express a preference, implementations need to be able to handle all reasonable values. There is no constraint that G.711 or other sample-based formats is conveyed in multiples of a certain unit. Thus, an RTP packet with 123 samples of G.711 is perfectly legitimate and needs to be handled appropriately.