SIP BasicsSIP FunctionalitySIP Protocol OperationRelationship to Other Protocols
SIP FAQ : SIP Basics
Answers in this category:
What is SIP?Who is implementing SIP?Where can I find more information about SIP?What is RFC2543bis?What is not described in the SIP spec?There are lots of SIP extensions. Will the protocol ever be
stable?
SIP FAQ : SIP Basics : What is SIP?
SIP FAQ : SIP Basics : Who is implementing SIP?
SIP FAQ : SIP Basics : Where can I find more information about SIP?
Until September 1999, discussion about SIP took place on the MMUSIC mailing list. The working group and mailing list is still the appropriate place for discussions related to SDP, SAP and RTSP, among other topics.
SIP standardization has now moved to the SIP working group, with its own mailing list. The SIP Forum at http://www.sipforum.org also provides information about SIP.
SIP FAQ : SIP Basics : What is RFC2543bis?
The latest version of this document can be found at http://www.cs.columbia.edu/~hgs/sip
SIP FAQ : SIP Basics : What is not described in the SIP spec?
- the color of the phones that use SIP;
- what operating system is used to implement a SIP phone;
- what language is used to display error messages;
- what sound, if any, is used to indicate ringback or ringing;
- whether a registrar, proxy or redirect server are implemented in one process, on one machine or distributed across a network;
- which media types or codecs the SIP implementation supports;
- which type of network SIP is used in;
- what type of security measures are required to be used;
- the GUI for SIP calls;
- how to build a network of proxy servers;
- whether a phone can make calls while it's in a call;
- qualifications for human beings to make use of SIP softphones;
- how generous an implementation should be in dealing with requests and responses that have mistakes and how much error detail information it should provide;
- how many simultaneous calls the device can be in.
- what parser generator tool, if any, should be used by implementations
- what threading model to use
SIP FAQ : SIP Basics : There are lots of SIP extensions. Will the protocol ever be stable?
Note also that just because something is proposed doesn't mean it will ever become a standard. (Only about 10% of Internet drafts ever get published as RFCs.)
SIP FAQ : SIP Functionality
Answers in this category:Does SIP support the standard telephone features?How does SIP provide message waiting services?Why does CSeq not have a compact form?How does the location server communicate with a proxy?How does SIP support caller ID?How do I charge/bill for Internet telephony using SIP?How do prepaid calling cards work in a SIP network?Does SIP do conference control?How do I put a call on hold?How does SIP do "call progress tones" or "ring back"?Does SIP do admission control?Does SIP administer bandwidth?How does SIP support multipoint conferences?How does SIP support text chat and instant messaging?Does SIP support video or multimedia conferences?When do I use a proxy server vs. a redirect server?
SIP FAQ : SIP Functionality : Does SIP support the standard telephone features?
- call forwarding unconditional, busy, ...
- call transfer (call control spec)
- caller ID
- call hold
- 3-way conferences and multiparty conferencing (call control spec)
- call return ("*69")
- call park (with NOTIFY)
- follow-me
- find-me
- call waiting
- IVR systems
- multiple line presences
- camp on
- call queueing
- automatic call distribution
- do not disturb
Some services, like repetitive dialing, station speed dialing, last number redial, and distinctive ringing, are implemented purely in the end system and require no support from the signaling protocol.
The Telecommunications Industry Association (TIA) is working on a recommendation http://www.tiaonline.org/standards/ip/ for business PBX-style services and other Internet phone requirements.
SIP FAQ : SIP Functionality : How does SIP provide message waiting services?
A SIP-based solution has not been standardized yet. Using SUBSCRIBE and
NOTIFY appears to be the most appropriate approach.
SIP FAQ : SIP Functionality : Why does CSeq not have a compact form?
SIP FAQ : SIP Functionality : How does the location server communicate with a proxy?
SIP FAQ : SIP Functionality : How does SIP support caller ID?
Since the callee generally does not know or trust the callee's server, only cryptographic signatures can be used to ensure that the information is valid. For example, the outgoing proxy might be operated by an ISP, enterprise or phone company and sign for the identity of the caller, using the signed-by parameter, with the identity of the company verified by a public key certificate similar to those used by web sites. See section 13.2 of RFC2543bis for encryption details.
SIP FAQ : SIP Functionality : How do I charge/bill for Internet telephony using SIP?
- SIP services
- The Authorization header can be used to indicate a
customer identity that associates a SIP request with a billable entity.
Examples of possibly chargeable SIP services include:
- Directory services such as SIP proxy/redirect lookups;
- Customer profile management;
- Media services
- Media services include retrieving and storing voice mail, as well as transcoding of media streams. They are not initiated by SIP, but, for example, via RTSP.
- Gateway services
- Similar to SIP services. Care has to be taken to stop billing when (say) RTP voice data is no longer flowing through the gateway. The gateway will generate call detail records (CDRs) either directly or through RADIUS.
- Transport (network services)
- It seems unlikely that voice calls carried over a best-effort
service will generate per-minute charges. When reserving bandwidth or
guaranteeing other quality-of-service parameters, the resource
reservation protocol or differentiated services are the appropriate
mechanism for including charging. These reservation protocols will
likely be used in applications that are not initiated by SIP, for
example, audio/video on demand or VPNs. Actual accounting records may be
generated by AAA protocols (e.g., by policy enforcement points (PEP) or
policy decision points (PDP)) or log files.
Under some circumstances, a SIP proxy server may be useful to initiate such reservations or differentiated services treatment on behalf of a call, since it may be easier to authenticate the SIP request than the lower-layer reservation request or the end system may not be capable of making reservations or marking packets. In those cases, the SIP proxy would initiate a resource reservation and "charge back" the caller identified by the SIP request.
Dean Willis wrote with regards to billing for SIP services:
Why can't service providers make a living providing (at a fixed cost) access to "free services"? Do carriers do per HTTP-transfer billing now? How much should they charge for an email? For a call, what parameters might be used? Bandwidth, duration, distance -- the Big Factors of the POTS bill -- are not issues that SIP is concerned with.
SIP FAQ : SIP Functionality : How do prepaid calling cards work in a SIP network?
SIP FAQ : SIP Functionality : Does SIP do conference control?
SIP FAQ : SIP Functionality : How do I put a call on hold?
SIP FAQ : SIP Functionality : How does SIP do "call progress tones" or "ring back"?
100 Message received 100 Looking up number 100 Found number, looking up carrier according to profile 100 Finding cheapest carrier which doesn't do animal testing 100 Found carrier "AT&T" 100 Dialing number 180 Ringing 182 Queued, 3 people in front of you 182 Queued, 2 people in front of you
The language of the status message should be determined based on the Accept-Language request header in the call.
A 183 (Session Progress) status response will appear in RFC2543bis. It can be used for both progress tones as well as error messages.
One would use the 183 only if you:
- Are able to determine that the audio being generated is something other than ringing (e.g. "comfort tone" or "pay tone" as defined in E.18x), or
- Are unable to definitively determine that alerting is occuring. This really should only happen with older CAS protocols. ISUP and ISDN have sufficient information to determine what is happening on the far end.
One can also use 183 if the gateway is able to determine that an error has occured, but that there is a tone or announcement accompanying it (e.g., an ACM with a cause code present). In that case, the gateway can send a 183 to set up the media for the announcement (ideally with the announcement text as the text string), wait for a timer (on the order of 30 seconds), and then send an appropriate SIP error message.
However, this should only be done if the caller is likely a human being, as sending 183 would otherwise only delay failure handling.
Take a look at (now expired) draft-ietf-sip-183-00.txt for some details on using 183 responses for early media announcements.
SIP FAQ : SIP Functionality : Does SIP do admission control?
SIP FAQ : SIP Functionality : Does SIP administer bandwidth?
There is a proposal for an SDP extension that allows SIP INVITE requests and responses to indicate that resource reservation must succeed before the callee is alerted. Several proposals can also be found at http://www.cs.columbia.edu/~hgs/sip/drafts_qos.html.
SIP FAQ : SIP Functionality : How does SIP support multipoint conferences?
1. Dialup conference bridge: call the bridge just like you call a person. Conference is identified by request URI. Works with rfc2543 - no extensions needed.
2. Distributed multiparty conferencing - no server. Fully distributed. This is what is described in the now-expired draft , and is work in progress (see http://www.softarmor.com/sipwg/teams/callcontrol/index.html for some recent drafts).
3. Multicast conferences - you can run your conference on multicast. Simply INVITE the person to join the multicast session. Works with baseline SIP. In fact, this was the initial purpose of SIP. In this case, there is not a full mesh of SIP signaling.
4. 3-way with local MC/MP function: A and B are talking. A wishes to add C to the call. A can simply call C, but also act as a mixer, so that the media it sends to B contains the A+C media, and the media to C is the A+B media. RTCP CSRC indicate who is in the conference. This also works with baseline SIP. It imposes additional burden on the UA, but otherwise provides this standard feature in a simple way.
There is no preferred way. It depends on the particular application. (Jonathan Rosenberg)
SIP FAQ : SIP Functionality : How does SIP support text chat and instant messaging?
Instant messaging (IM) differs from text chat in that:
- messages are sent as a whole and can be in any format, such as HTML;
- messages are independent of a session setup;
- store-and-forward or translation to email are feasible;
SIP can support IM through the use of the proposed MESSAGE method.
SIP FAQ : SIP Functionality : Does SIP support video or multimedia conferences?
- dial-in-style conferences with a central "bridge" (MCU) or multicast
- dial-out conferences with a central bridge or multicast
- multicast conferences
- full or partial mesh conferences, where end systems replicate media streams.
SIP FAQ : SIP Functionality : When do I use a proxy server vs. a redirect server?
You use a proxy when you want to control processing of the call from the point you receive it, and forward. A proxy can see the provisional and final responses to the request. It can record route so that it sees whats going on during the call. A redirect server, however, hands off control to the device that sent it the INVITE. It will never see the final response to the request, and not be contacted again for the remainder of the call. Its a one-shot deal.
As such, redirect servers are really good for high volume, lookup style transactions. They can almost be considered a form of database query, albeit a SIP specific one. Proxies are generally needed for services and more complex routing problems.
2000-Nov-14 10:50am mailto:jdrosen@dynamicsoft.com?subject=SIP FAQ
SIP FAQ : SIP Protocol Operation
Answers in this category:What does the [H14.17] in RFC 2543 stand for?Do callers need to know the location of the Location
Server?What is the difference between a call leg and a call
id?What is the difference between tag and branch-id?How can one recognize a retransmitted, duplicate or looped
request?What is the relationship between the From, Contact, Via and
Record-Route/Route headers?How are URLs compared?What's the difference between the request URIs tel:+12125551212
and sip:12125551212@gw.com?Do I always need a proxy or redirect server?How does a caller find its local registrar?How do I ensure registrar reliability?Are ACK requests retransmitted?How are BYE requests routed?Can I CANCEL requests other than the first INVITE?How does a caller find its proxy server?What's the difference between a stateless and a stateful proxy
server?Why can a forking SIP proxy not be stateless?How does a caller find the remote SIP client of the
callee?How does SIP get through a firewall?Does SIP do keep-alive?Why does SIP not have a Content-Transfer-Encoding
header?I want SIP to be more compact. What can I do?What are the different addresses in SIP?The BNF for header <put your favorite header here> allows a
parameter to appear more than once. What does this mean?Can the request URI include a port number and/or transport
parameter?Transport in ViaWhat should I do if my re-INVITE fails?How long can SIP host names be?Can a User Agent also act as a Registrar?Can I remove an m= line from SDP in response or
re-INVITE?I'm a UAC. I sent an INVITE, and then decide I want to hang up
before getting a final response. Do I send BYE or CANCEL?I'm a proxy, and I forked a request, and forwarded multiple 200
OK upstream. Now, I get an ACK. What do I do with it?If I get a new SDP body in the ACK, and I don't like the media
type, how can I indicate its unacceptable to me?Can a SIP UA register with multiple registrars?Is it possible for a UA to make a call to itself, and have the
result be two separate calls on the same machine?Does a UAS use the request-URI or To field to determine if a call
is for it?How are SIP parsers implemented?Is it possible to use Hide with Record-Route?How does a proxy handle a method other than the standard INVITE,
ACK, BYE, etc.?Why does a proxy server doing TCP need to be stateful?In computing the Content-Length, does the newline in a body count
as one byte or two (CR vs. CR-LF)?Can a proxy fork a non-INVITE request? If yes, what happens if it
gets multiple responses?Should responses be sent to the host specified in Via? Is From
ever used for sending responses?Once a SIP registrar gets a REGISTER request, how does it update
the Location Server with the contact information?Is a SIP URI without a user name valid?Is there a specific order for header fields?When is a CANCEL used?What do I need to do to use SRV records?
SIP FAQ : SIP Protocol Operation : What does the [H14.17] in RFC 2543 stand for?
SIP FAQ : SIP Protocol Operation : Do callers need to know the location of the Location Server?
- finger;
- LDAP/X.500;
- whois, whois++;
- ph and other local directories;
- shared file systems with registration on login;
- local SQL databases reached through TCP.
Also, callers don't register with the location server.
SIP FAQ : SIP Protocol Operation : What is the difference between a call leg and a call id?
Call legs refer to end-to-end connections between user agents, rather than any relationship with proxies. Within a call leg, there are numerous transactions in both directions.
The request URI is not used in call leg identification.
The To and From field relate to local and remote in the following way. When Alice sends a request on a call leg to Bob, the From field contains the local address (Alice), and the To field the remote address (Bob). When a request is received by Bob, the To field is matched to Bob's local address, and the From field to the remote address (Alice).
The CSeq spaces in the two directions of a call leg are independent.
Within a single direction, the sequence number is incremented for each
transaction.
SIP FAQ : SIP Protocol Operation : What is the difference between tag and branch-id?
A UAS has no reliable way of determining if the request has been forked or not. Thus, to be safe it needs to add a tag. Proxies only insert tags into the final responses they generate themselves; they never insert tags into requests or responses they forward.
Since a request can be forked several times on its way to UAS, a single
"tag" (or whatever you like to call it) added to the request by one of the
proxies is not sufficient for the next forking proxy along the chain to
match responses on its own branches; every proxy that forked the request
would need to add its own unique IDs to the branches it created. This is
precisely what's being achieved by the branch parameter in the Via header.
SIP FAQ : SIP Protocol Operation : How can one recognize a retransmitted, duplicate or looped request?
header retransmitted duplicate matching response From same same same To same same same, but tag may have been added Call-ID same same same request URI same same n/a CSeq same same same Via - - must be local host; check for branch parameter to identify which branch
Looped request are recognized by one of the following:
- The server finds itself in the request's Via list, including any branch parameter. (The server should compute the branch parameter so that it depends on the request URI.) Take a look at sections 6.47.5 and 12.13.1 of RFC2543bis for details on computing and matching branch parameter for loop detection.
- The Via header added by the server prior to proxying the request is the same as one of the Via's in the Via list.
SIP FAQ : SIP Protocol Operation : What is the relationship between the From, Contact, Via and Record-Route/Route headers?
- From:
Used for subsequent requests from the callee to the caller
if there is no Contact or Record-Route header. E.g., if Alice makes a call
with
From: Alice sip:alice@example.orgto Bob, an INVITE request from Bob to Alice would use
sip:alice@example.orgas the To header and Request-URI.
- Contact:
Determines the destination placed in the Request-URI for
subsequent requests and can be used to bypass proxies _not_ enumerated in
a Record-Route header. Also used in responses by redirect servers and in
REGISTER requests and responses.
- Record-Route/Route:
The Record-Route header is inserted into
requests by proxies that want to be in the path of subsequent requests for
the same call-id. It is then used by the user agent to route subsequent
requests. The mechanism is similar to a source-route, as the Record-Route
information is copied into a set of Route headers. The Request-URI is set
to the first Route header.
- Via:
Via headers are inserted by servers into requests to detect
loops and to allow responses to find their way back to the client. They
have no influence on the routing of future requests (or responses).
Generally, in short, requests should be sent to Route if present,
Contact if there is no Route, From if there is no Contact.
SIP FAQ : SIP Protocol Operation : How are URLs compared?
- the display name is ignored;
- tags must match;
- user, password, host, port and parameters of the URI must match. If a component is omitted, it matches based on its default value.
- string comparisons are case-sensitive except for the domain part;
- Characters other than those in the "reserved" and "unsafe" sets (see RFC 2396) are equivalent to their ""%" HEX HEX" encoding.
- An IP address that is the result of a DNS lookup on a hostname does
not match that hostname.
SIP FAQ : SIP Protocol Operation : What's the difference between the request URIs tel:+12125551212 and sip:12125551212@gw.com?
SIP FAQ : SIP Protocol Operation : Do I always need a proxy or redirect server?
SIP FAQ : SIP Protocol Operation : How does a caller find its local registrar?
Another possibility is to use SLP through an extensions defined in http://www.cs.columbia.edu/~hgs/sip/drafts/draft-kempf-sip-findsrv-00.txt.
SIP FAQ : SIP Protocol Operation : How do I ensure registrar reliability?
- Run several servers that all respond to the same multicast registration address ("warm standby"). As long as multicast requests are mostly reliable, this ensures a consistent registration picture.
- If a registration server is rebooted and does not have complete knowledge of the local UA population, it could multicast any incoming INVITE requests.
- Use some of the available hardware redundancy solutions.
For servers separated from their client by a wide-area network, use of
multicast is not appropriate, so that these servers have to rely on
traditional backup techniques to achieve reliability. For example, the
designated registrar could multicast registration updates within its local
network to keep standby servers synchronized.
SIP FAQ : SIP Protocol Operation : Are ACK requests retransmitted?
SIP FAQ : SIP Protocol Operation : How are BYE requests routed?
If the caller decides to send a BYE before receiving a 200 from the
callee, the BYE is be handled by the proxies just as the corresponding
INVITE was handled, i.e., it may be forked.
2000-Jul-03 9:21pm mailto:islepchin@dynamicsoft.com?subject=SIP
FAQ
SIP FAQ : SIP Protocol Operation : Can I CANCEL requests other than the first INVITE?
SIP FAQ : SIP Protocol Operation : How does a caller find its proxy server?
If a "local" (outbound) proxy is needed for outgoing calls, it
currently needs to be manually configured, similar to the configuration of
web proxies in browsers. Another possibility is to use DHCP or SLP through
one of the extensions listed in http://www.cs.columbia.edu/~hgs/sip/drafts_dhcp.html.
SIP FAQ : SIP Protocol Operation : What's the difference between a stateless and a stateful proxy server?
Stateless proxies scale very well, and can be very fast. They are good for network cores. Stateful proxies can do more (they can fork, for example, see the next question) and can provide services stateless ones can't (call forward busy, for example). They don't scale as much as stateless ones. An admininstrator gets to decide which to use. These are also logical entities; a physical proxy is likely to act as a stateless proxy for some calls, stateful for others, and as a redirect server for even others.
Neither stateful nor stateless proxies need to maintain call state, although they can, but will need to make sure that they are part of subsequent transactions via the Record-Route header.
A proxy must be stateful if one of the following conditions hold:
1. It uses TCP,
2. It uses multicast,
3. It forks.
SIP FAQ : SIP Protocol Operation : Why can a forking SIP proxy not be stateless?
Also note that a proxy that uses TCP must be stateful as well, whether it forks or not. This has to do with reliability issues.
Why do you want state in a proxy? Certain services (like forking)
simply require it. A sequential search proxy requires state; sequential
search is the heart of services like follow-me and personal mobility. It's
at the discretion of the implementor whether to use a stateful or
stateless proxy. You can even be "super stateful", and use the
Record-Route header to allow a proxy to be on the signaling path of all
subsequent exchanges. This allows a stateful proxy to maintain call state
in addition to transaction state.
SIP FAQ : SIP Protocol Operation : How does a caller find the remote SIP client of the callee?
sip:bell@cs.columbia.edu,the client finds a SRV record pointing to
erlang.cs.columbia.eduas the SIP server for the domain cs.columbia.edu. As for email, a single domain name can resolve to multiple servers, allowing load sharing and redundancy.
The server located in this manner can then proxy or forward the call to
another server.
SIP FAQ : SIP Protocol Operation : How does SIP get through a firewall?
As an alternative, if a firewall or NAT allows outgoing TCP connections, the inside client can open up a TCP connection to an outside proxy. All outgoing and incoming calls would then be handled by that TCP connection. (The client would still have to use SOCKS or similar mechanism to convince the firewall to let RTP packets through.)
Take a look at the two dratfs at http://www.cs.columbia.edu/~hgs/sip/drafts_firewall.html
for a more detaled discussion of getting SIP through firewalls and
NATs.
SIP FAQ : SIP Protocol Operation : Does SIP do keep-alive?
However, knowing the call state might be useful for certain
applications (e.g., when billing is involved, when firewall permissions
need to be set etc.). Session timer extension has been defined to solve
this. The draft can be found at http://www.cs.columbia.edu/~hgs/sip/drafts/draft-ietf-sip-session-timer-01.txt
and it basically allows the servers indicate a desired refresh interval.
The call is considered terminated if a re-INVITE is not received within
that interval.
SIP FAQ : SIP Protocol Operation : Why does SIP not have a Content-Transfer-Encoding header?
See also RFC 2616 (HTTP/1.1), Section 19.4.5.
SIP FAQ : SIP Protocol Operation : I want SIP to be more compact. What can I do?
In general, more substantive savings are possible by using either
payload compression (RFC 2393) or link-layer compression, e.g., at the PPP
layer. For the example above, the total size is reduced to about 520 bytes
with gzip compression.
SIP FAQ : SIP Protocol Operation : What are the different addresses in SIP?
1.The host address where the request came from. Responses are sent back to the same host address, regardless of what the From header indicates. Note that different requests for the same call can come from different hosts.
2.The From address contains the logical source of the request. It remains unmodified as a SIP request traverses proxies, for example. The From address may not be the same as the host address that generated the SIP request, although that's the typical case.
3.The session description (e.g., SDP) contains one or more addresses
where the caller expects media data (audio, video) to be sent. For some
services, this address may not be the same as the From
address.
SIP FAQ : SIP Protocol Operation : The BNF for header <put your favorite header here> allows a parameter to appear more than once. What does this mean?
SIP FAQ : SIP Protocol Operation : Can the request URI include a port number and/or transport parameter?
Lets say I send a request to joe@example.com, and the server for example.com is listening on 5061. The request URI might look like:
INVITE sip:joe@example.com:5061 SIP/2.0
this arrives at example.com. Since the request is for that server, it looks up joe in some database and translates the request URI (for example, to sip:joe@engineering.example.com). It looks up engineering.example.com in DNS, and finds an A record for that machine, forwarding the request to the given IP address. The outgoing request URI looks like:
INVITE sip:joe@engineering.example.com SIP/2.0
Note that in this case, the presence of port 5061 in the request URI sent to example.com didn't make a difference. Thats because the example.com just translated the request URI. Whether it had contained the port number or not would have had no effect on processing.
However, had the request instead been sent to a local outbound proxy instead of example.com, the port number would NEED to be there. Thats because the local outbound proxy won't translate the request URI, it will example it, determine its not for itself, look up the domain in the request URI in DNS, and forward the request there. So, the request URI needs to contain this port so that the local outbound proxy knows to forward it to 5061 as opposed to 5060 at example.com.
So, the rule of thumb is this:
if you send a request to the server listed in the domain of the request URI, URI parameters like port, transport, ttl etc MAY be present but are not needed. If you send a request to a server which is NOT the one identified in the request URI, you MUST include these parameters if they are not the defaults. Always inclduing them, when not default, means you don't need to determine which is the case, and is always the safest bet.
SIP FAQ : SIP Protocol Operation : Transport in Via
If such a request is received, it should be responded to with a 400 Bad
Request. The protocol used for the request should also be used for the
response.
SIP FAQ : SIP Protocol Operation : What should I do if my re-INVITE fails?
SIP FAQ : SIP Protocol Operation : How long can SIP host names be?
SIP FAQ : SIP Protocol Operation : Can a User Agent also act as a Registrar?
SIP FAQ : SIP Protocol Operation : Can I remove an m= line from SDP in response or re-INVITE?
The reason for this is that we need to ensure that it is always possible to match media sessions (i.e., "m=" lines) in requests and responses. Consider an INVITE with the following SDP:
... c=IN IP4 1.2.3.4 m=audio 54678 RTP/AVP 0 1 3 m=video 7346 RTP/AVP 28 31 (face) m=video 7880 RTP/AVP 26 28 (presentation)
If the response contained something like
... c=IN IP4 3.4.5.6 m=audio 6540 RTP/AVP 0 1 m=video 6578 RTP/AVP 28
the caller would not be able to tell which of the two offered video
streams was accepted.
SIP FAQ : SIP Protocol Operation : I'm a UAC. I sent an INVITE, and then decide I want to hang up before getting a final response. Do I send BYE or CANCEL?
Sending CANCEL helps solve many of these problems. CANCEL will reach the same set of recipients as the original INVITE, and it doesn't need a REcord-Route or tag in it. The drawback, however, is that the CANCEL and a 200 OK from one of the UAS might pass on the wire. Thus, the UAC may still need to ACK the 200 OK, and then send BYE. The other drawback is that you wouldn't send CANCEL if the call was already established, you'd send BYE. Folks complained they didn't want to have state-dependent mechanisms for hanging up. Given the unlikelihood of the problems with sending BYE, it seems reasonable to allow it.
SIP FAQ : SIP Protocol Operation : I'm a proxy, and I forked a request, and forwarded multiple 200 OK upstream. Now, I get an ACK. What do I do with it?
That aside, should it arrive anyway, the ACK should be routed just as any other new request. Apply routing logic, which presumably causes it to be forked to both locations. The tags will help identify for which UAS the ACK is meant.
SIP FAQ : SIP Protocol Operation : If I get a new SDP body in the ACK, and I don't like the media type, how can I indicate its unacceptable to me?
SIP FAQ : SIP Protocol Operation : Can a SIP UA register with multiple registrars?
SIP FAQ : SIP Protocol Operation : Is it possible for a UA to make a call to itself, and have the result be two separate calls on the same machine?
If you have a table of calls by Call-ID only, you will run into problems as you may think this is a re-INVITE for the same call. You really want the incoming and outgoing messages to be associated with totally different calls. It can be distinguished as a separate call if, again, you allow for direction in your lookups. There are many ways to do this; the easiest is to include your local address as part of the call key. For incoming requests, the key is callID + from, for outgoing, callID+ to.
SIP FAQ : SIP Protocol Operation : Does a UAS use the request-URI or To field to determine if a call is for it?
SIP FAQ : SIP Protocol Operation : How are SIP parsers implemented?
- http://www.gnu.org/software/flex/flex.html flex for lexical analysis
- bison (http://www.gnu.org/software/bison/bison.html) or lemon (http://www.hwaci.com/sw/lemon/) for parsing.
A somewhat outdated grammar summary can be found at http://www.cs.columbia.edu/~hgs/sip/SIPgrammar.html
SIP FAQ : SIP Protocol Operation : Is it possible to use Hide with Record-Route?
SIP FAQ : SIP Protocol Operation : How does a proxy handle a method other than the standard INVITE, ACK, BYE, etc.?
SIP FAQ : SIP Protocol Operation : Why does a proxy server doing TCP need to be stateful?
TCP UDP UA1 -------- Proxy --------- UA2
Lets assume that this proxy is stateless, meaning that it holds the TCP
connection state but otherwise does not store transaction state. According
to the specification, requests are not retransmitted over TCP. So, the UA1
sends its request, say an INVITE, just once over the TCP connection. The
proxy receives this, and forwards it to UA2. Its lost. UA1 will not
retransmit (since its TCP), and neither will the proxy, since its
stateless, thus, the message is lost.
SIP FAQ : SIP Protocol Operation : In computing the Content-Length, does the newline in a body count as one byte or two (CR vs. CR-LF)?
SIP FAQ : SIP Protocol Operation : Can a proxy fork a non-INVITE request? If yes, what happens if it gets multiple responses?
As a result of this, forking of non-INVITE requests is only useful when the method has semantics that meet certain criteria. Specifically, (1) the client doesn't care which server gets the request, (2) the client doesn't care which server sent the response, or even if multiple servers sent a response, (3) the service provided by each server is identical. In essence, forking of non-INVITE requests is useful only for an anycast type of service.
SIP FAQ : SIP Protocol Operation : Should responses be sent to the host specified in Via? Is From ever used for sending responses?
Note that if you are a UAS, you are the entity responsible for adding
"received" parameter to the top Via so you may want to interpret the last
two rules in the section as requiring to send the response to the source
address of the request.
SIP FAQ : SIP Protocol Operation : Once a SIP registrar gets a REGISTER request, how does it update the Location Server with the contact information?
SIP FAQ : SIP Protocol Operation : Is a SIP URI without a user name valid?
SIP FAQ : SIP Protocol Operation : Is there a specific order for header fields?
HTTP/1.1 says:
"The order in which header fields with differing field names are received is not significant."
"The order in which header fields with the same field-name are received is therefore significant to the interpretation of the combined field value, and thus a proxy MUST NOT change the order of these field values when a message is forwarded."
If a set of fields is authenticated, proxies must not re-order or otherwise modify these field, as this would break the authentication.
SIP FAQ : SIP Protocol Operation : When is a CANCEL used?
- The time described in the Expires header of the request has elapsed;
- No response, including provisional, was ever received from downstream nodes;
- Internal logic determines its time to end the transaction (a CPL or sip-cgi script, for example).
SIP FAQ : SIP Protocol Operation : What do I need to do to use SRV records?
_sip._tcp SRV 0 0 5060 sip-server.cs.columbia.edu.
SRV 1 0 5060 backup.ip-provider.net.
_sip._udp SRV 0 0 5060 sip-server.cs.columbia.edu.
SRV 1 0 5060 backup.ip-provider.net.
DNS SRV
records are supported by BIND 4.9.6 and newer, generally installed as
named. Configuring named for Linux is discussed in a HOWTO at http://www.linuxdoc.org/HOWTO/DNS-HOWTO.htmlCurrently registered SRV records:
sip.tcp.cs.columbia.edu SRV 0 0 5060 erlang.cs.columbia.edu sip.udp.sip-happens.com SRV 0 0 5060 sip.sip-happens.comUse, for example,
host -v -t srv sip.tcp.cs.columbia.edu host -v -t srv sip.udp.cs.columbia.edu host -v -t srv _sip._udp.cs.columbia.edu
SIP FAQ : Relationship to Other Protocols
Answers in this category:What is the relationship between MGCP and SIP?What is SIP+ and how does it relate to SIPHow does SIP compare to H.323?Can H.323 and SIP be used together? How do I interconnect Q.931 (ISDN signaling) and SIP?How do I interconnect ISUP (SS7 signaling) and SIP?Where do I find description of SDP?Can SIP be used for Internet telephony gateways (ITGs)?What is sip-cgi and how does it relate to CPL?Is there a SIP interoperability certification? How can I test
interoperability with others?Why use SIP-T as opposed to tunneling SS7 using SCTP?How does SIP carry DTMF (touch tones)?What is the relationship of a "softswitch" to SIP?How does SIP/SDP relate to T.38 fax calls?Where is the use of SIP defined in 3GPP?
SIP FAQ : Relationship to Other Protocols : What is the relationship between MGCP and SIP?
Additional details provided by Tom Taylor:
sip_h323_mgcp.gif (28.1 K)The basic architecture assumed by the Megaco Working Group postulates two functional entities: a Media Gateway Controller (MGC), which owns the call model and is responsible for call signalling, and a Media Gateway (MG), responsible for manipulating (directing, transforming) media flows under the control of the MG.
MGCP and Megaco/H.GCP are both protocols used between the MGC and MG when they are realized in separate physical elements. MGCP (Media Gateway Control Protocol) was a major source of the ideas in the current Megaco/H.GCP protocol draft, and is being deployed in a number of products being announced over the next few months. It is best suited for IP telephony gateway applications. The Megaco protocol is also called H.GCP because it is being developed cooperatively between the Megaco WG and ITU-T Study Group 16.
H.323 is a complete system specification, including call signalling
protocols which would run between an MGC and another MGC or other H.323
entities (Gatekeepers, endpoints). SIP can also be used as a call
signalling protocol, and can therefore be viewed as a competitor to H.323.
Both protocols are capable of supporting multipoint multimedia
conferences. H.323 was first standardized in 1996 and has been improved
since then; current standardization is focusing on networking aspects such
as translations data exchange and interworking with legacy telephony
signalling. SIP just reached Proposed Standard status, but has attracted
wide interest which may speed its maturing stages. The Megaco/H.GCP
protocol will complement both protocols by also providing support for
multipoint, multimedia calls at the media level.
SIP FAQ : Relationship to Other Protocols : What is SIP+ and how does it relate to SIP
SIP FAQ : Relationship to Other Protocols : How does SIP compare to H.323?
It also supports a number of services, such as ACD and follow-me, that are much harder to implement in H.323.
SIP FAQ : Relationship to Other Protocols : Can H.323 and SIP be used together?
Unfortunately, there is currently no standard on translating between the two. Conversion is made more difficult by the multiple versions of H.323 (v1, v2, v3). However, there is at least one product (Lucent PacketStar IP) that allows SIP and H.323 terminals to call each other.
There is an ongoing effort within SIP Working Group to define SIP-H.323
interoperation standard. Some details on the effort are available at http://www.softarmor.com/sipwg/teams/sip-h323/index.html.
The group has produced an internet dratf that can be found at http://www.cs.columbia.edu/~hgs/sip/drafts/draft-singh-sip-h323-00.txt.
SIP FAQ : Relationship to Other Protocols : How do I interconnect Q.931 (ISDN signaling) and SIP?
isdn_sip.pdf (8.7 K)
Some additional information and links to related Internet Drafts can be
found at http://www.softarmor.com/sipwg/teams/sipt/index.html.
SIP FAQ : Relationship to Other Protocols : How do I interconnect ISUP (SS7 signaling) and SIP?
sip_isup.pdf (7.0 K).
Some additional information and links to related Internet Drafts can be
found at http://www.softarmor.com/sipwg/teams/sipt/index.html.
SIP FAQ : Relationship to Other Protocols : Where do I find description of SDP?
SIP FAQ : Relationship to Other Protocols : Can SIP be used for Internet telephony gateways (ITGs)?
SIP FAQ : Relationship to Other Protocols : What is sip-cgi and how does it relate to CPL?
CPL is an XML-based language, while sip-cgi is a mechanism for invoking scripts or programs written in any language. sip-cgi is very similar to web cgi scripts.
In its current version, CPL is only invoked when INVITE requests and responses arrive, while sip-cgi can intercept any request.
sip-cgi is designed to be used by SIP, while CPL can probably be used by a number of signaling protocols such as Q.931 or H.323.
CPL and sip-cgi differ in their applicability. CPL is designed for end user service creation. It is intentionally limited in capabilities and is not a general purpose programming language. Its execution on a server is generally very fast. CGI is more powerful - you can do nearly anything. It is programming language independent. It incurs a process-spawning overhead, so its less efficient than CPL. (CPL is usually executed in the same process as the server). As a service provider, I would not want to execute CGI scripts sent to me by end users. However, I would prefer to use CGI to develop my own services.
Note that CGI may be used as the execution environment for a CPL
script. (Jonathan Rosenberg)
SIP FAQ : Relationship to Other Protocols : Is there a SIP interoperability certification? How can I test interoperability with others?
SIP FAQ : Relationship to Other Protocols : Why use SIP-T as opposed to tunneling SS7 using SCTP?
SIP FAQ : Relationship to Other Protocols : How does SIP carry DTMF (touch tones)?
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.
SIP FAQ : Relationship to Other Protocols : What is the relationship of a "softswitch" to SIP?
SIP FAQ : Relationship to Other Protocols : How does SIP/SDP relate to T.38 fax calls?
SIP FAQ : Relationship to Other Protocols : Where is the use of SIP defined in 3GPP?
The draft version for TS24.228 is available at http://www.3gpp.org/ftp/Specs/2000-12/Drafts/24_series/