Access This refers to how you route the SIP calls on the LAN side of your network. For example, broadband, Ethernet or a private interconnect.
ADSL
(Asymmetric Digital Subscriber Line)
The majority of SIP trunks will route over an ADSL (Broadband) connection. ADSL enables faster transmission over traditional copper telephone lines. A micro-filter is installed on the end user's phone line to enable both ADSL and traditional voice services to be used at the same time.
ALG
(Application Level
Gateway)
SIP application-layer gateways can be found in many routers and firewalls. When implemented correctly, they help to resolve NAT-related issues, inspecting SIP traffic and re-writing private, internal IP addresses with public, routable IP addresses.
Anonymous See the image below for an example of a call trace whereupon the caller wishes to
withhold their CLI. If the CLI in the 'from' header is overwritten with 'Anonymous', a valid privacy header needs to be sent in the P Asserted Identity; this enables us to pass a valid network number to onward carriers, for the purposes of compliance and emergency dialling.
B Numbers B-numbers relate to 'called party' or 'destination' numbers, and should be sent to us in the following format:
UK national 0+NSN (national significant number) - 01418701234
+44+NSN - +441418701234
International 00+CC+NSN - 00441418701234
Service and emergency calls no leading 0 or CC (country code)
As a default configuration B-numbers will be presented to the customer including a leading 0.
When reporting a SIP fault to us, you will be asked to supply time-stamped examples of
B numbers to which calls have failed.
Call
Admission Control (CAC)
Through a process known as Call Admission Control (CAC), the maximum call limit of an endpoint defines its capacity for routing calls in the network. SIP trunking customers pay a fixed monthly charge for the number of concurrent calls allowed on their
Thus we will support any combination of incoming or outgoing calls provided the total number of calls does not exceed the total channel allocation (i.e. CAC limit).
Call Barring By default, calls to international numbers will be barred. For automated endpoints, customers can modify their profiles, to enable both international and premium rate dialling. Similarly, all call barring can be applied in the event of fraudulent activity on an endpoint. Calls to emergency services will remain unaffected irrespective of the barring in place.
Call Transfer We do not support the REFER method for call transfers. If the parameters of an
existing SIP session are to be modified, this will be achieved by a re-INVITE, with C-Sequence incrementation as part of the same call.
Calls per Second (CPS)
Maximum Calls per Second (CPS). We set limitations to CPS, and will reject calls with a '486 busy here' response, should this limit be reached. Typically, CPS limit is set to 2 CPS for SIP trunks with up to 30 channels, and 5 CPS for endpoints with more than 30 channels.
CLI Flexibility
Our SIP Trunking service by default supports outbound presentation from a PBX within the following format:
If the PBX presents a geographic number, our Network will relay its details as the A-Number CLI into the PSTN or Mobile networks, providing that this number has been allocated to the endpoint, either at the time the order was placed, or subsequent to a number port.
CLI Flexibility enables the presentation of non-Gamma registered CLIs in the 'from' header of your SIP INVITE. This is contingent upon sending a valid network CLI in the PAID (P Asserted ID) header. The presence of a network CLI enables its association to physical premises in compliance with OFCOM regulations.
For any calls made to the Emergency Services, we will automatically present the default CLI, which is the first number in the Gamma allocated DDI range.
We will not support the presentation of non-UK (international) or 09 numbers
CLIP Calling Line Identification Presentation (CLIP) is a service that transmits a caller's number to the called party's telephone equipment during the ringing signal.
Codecs Codecs are used to convert analogue voice signals into digital data for transmission
across an IP network.
Our default codec for IPDC is G711alaw, though we also support G711ulaw and G729 codec.
Codecs vary in sound quality, with G729 the preferred codec when access is via ADSL, owing to its compressed nature and efficient use of bandwidth.
The maximum number of concurrent channels = the available bandwidth/total
bandwidth, so if for example a customer has a 512 kbps upload line-speed, assuming no contention, using G729 with a sample period of 20ms there will be 512/40 ≈ 10 usable channels available.
Contact Header
A field in the SIP INVITE in which a User Agent (SIP device) specifies an IP address and port, to which subsequent responses are to be sent. The IP address should be the public address of the endpoint and the port, in most cases 5060, the default port for SIP signalling.
Customer Premises Equipment (CPE)
The provisioning and support of all CPE (Customer Premises Equipment) at the end user site is the responsibility of the customer. We undertake regular conformance testing with equipment vendors to ensure compatibility of equipment and ease of installation. A list of compatible CPE can be provided upon request.
Delay Each component in the path adds delay (sender, network, and receiver). ITU-T G.114
recommends 150ms as maximum desired delay to achieve high voice quality.
Diverts Under failure conditions, customers can contact us so we can apply diverts, either to
single numbers or DDI ranges. Please be aware that diverts can take several moments to become effective, as the information is written to back-end routing nodes.
Diverted destinations are subject to the same call barring options as the main SIP trunk, e.g. if the user does not allow calls to mobiles, then the divert destination options will also exclude mobile numbers.
DTMF DTMF stands for Dual Tone - Multi Frequency and your touch-tone® phone is
technically a DTMF generator that produces DTMF tones as you press the buttons. We support the following methods for the transport of DTMF tones:
RFC2833, or 'out of band', is the preferred method for the DTMF transit. RFC2833 requires payload 101 to be assigned.
In-band DTMF transmission is also supported, and can be used with G711 or G729 codec (though detection of tones at the far end is not guaranteed when G729 is in use).
We do not support the transmission of DTMF events as part of a SIP INFO message, and will issue a 415 'unsupported' response.
DUAL ENDPOINT
If our customer's equipment is unable to implement near-end NAT (the mapping of private IP addresses to routable, public IP addresses), we may be able to build a Dual Endpoint to circumvent the translation. This entails the implementation of a 'far-end' NAT, whereupon our Session Border Controller is configured to receive SIP headers containing private IP addresses from a particular endpoint. Rather than issuing a 403 forbidden message upon receipt of an INVITE, the Session Border Controller performs a NAT translation. Please note that Dual Endpoints can only be commissioned in certain circumstances.
Emergency 999 Calls
This is a VOIP service as defined by OFCOM and can be used to support Emergency Services calls. Once the service is fully operational, 999/112 public emergency call services can be accessed and will be routed to the national emergency call handling agents. The CLI presented will always be the site CLI, indicated as a VOIP service type from Gamma, so that the emergency services operator can verify the address details.
Emergency dialling may not be possible in the following circumstances:
During a power outage or the failure of DSL routing equipment.
If an end user’s account has been suspended
In such circumstances the end-customer should use their PSTN line to make the emergency call.
In addition, the end-user should also be made aware that the emergency personnel would need to confirm the identity and the actual location of the caller when they dial 999/112.
Customers should always be informed of the above service limitations relating to the Emergency Services support in line with the OFCOM Code of Practice.
Endpoint The end user's CPE is defined as an ‘endpoint’ within our Session Border Controller.
The endpoint will be setup with a public IP address and a pre-defined number of channels.
Fax The following FAX methods are supported G711 pass-through for 'in-band' transmission.
T.38 relay, within which either party issues a SIP re-INVITE with T.38, upon the detection of fax tones.
Invite SDP
IP
Addressing
We do not yet support IP version 6 addressing
IPDCv3.0/3.1 SIP username and password registration. The customer's PBX will send registration packets from its public IP address, to which we will route calls.
IPDCv3 services are authenticated by password, transmitted by the PBX in response to a 407 challenge issued by our SIP proxy. SIP registrar is a legacy product, and we will not be provisioning further services upon this platform.
Jitter Jitter is variation in packet delay. Greater levels of jitter are likely to occur on either slow or heavily congested links. If you are experience call quality issues, we can add your public IP address to its Smokeping platform, which will accrue statistics as to any jitter, latency or packet loss apparent. Where jitter is evident, a Smokeping graph will appear as outlined
Local Dialling
Gamma does not support local dialling across its SIP trunks
Long Calls We will, in most instances, clear down 'long hold' calls that exceed eight hours in duration.
Network CLI Every endpoint must have at least one CLI from the Gamma-allocated range i.e. a
non-ported in number. This default number is known as the Network CLI and will be presented in the case of emergency calls. The network CLI must be associated to a physical premises address. For onward delivery to PSTN or mobile networks, the network CLI will be mapped from the SIP P Asserted ID or Remote Party ID in your INVITE.
Packet Loss Packet loss is the failure of transmitted packets to arrive at their destination. The presence of packet loss is generally indicative of a problem on an access link. We can add your public IP address to its Smokeping platform, which will accrue statistics as to any jitter, latency or packet loss apparent. Where packet loss is evident, a Smokeping graph will appear as outlined
Packetisatio n
Packetisation delay is the time taken to fill a packet payload with encoded speech. This delay is a function of the sample block size required by the vocoder and the number of blocks placed in a single frame. Packetisation delay is also known as accumulation delay, as the voice samples accumulate in a buffer prior to their release. We offer both 10ms and 20ms packetisation rates.
Presentation CLI
The FROM header in your SIP INVITE should be denoted without the leading zero. This A-number will be validated by our Session Border Controller, and overwritten with the base number of the SIP trunk, if there are no CLI verification rules in place to enable the A-number to present.
The P Asserted ID should be presented in E.164 format (+441618703358); deviating from this format can result in number presentation problems in certain call flows.
Privacy Header
See the image below to see an example of a valid privacy header, sent in the form of a P Asserted Identity header, where the originating CLI in the 'from' header has been withheld.
Private Interconnect
In order to nullify security concerns, we offer improved access options:
Private interconnection via our IP Assured service
Private Interconnect Access via the Gamma network
Access via private interconnection is possible at the following Gamma peering points:
Telicity 2, 8/9 Harbour Exchange Square
Telehouse East, E14 2AA
Redbus, 6/7 Harbour Exchange Square
Telehouse North, E14 2AA
Epsilon Global Hubs, London
City Lifeline, 80 Clifton St, London,
Telecity Powergate, London
Kilburn House at Telecity, Manchester
Connection Types include copper, single or multi-mode fibre and bandwidths ranging from 10Mbps to 1Gbps. Interconnect port types include Layer 3 BGP, Layer 2 VLAN and Layer 2 802.1q VLAN. For improved redundancy resilience, resilient connections should be considered rather than standalone builds.
REQUEST-URI
The Request-URI is the intended destination of the SIP INVITE request (the UA the sending device is trying to contact).
RTP Real Time Transport Protocol opens up a port for the transport of media (speech). Our media gateway addresses are specified in the e-mail provided by our IP Provisioning team, upon an endpoint's completion. Typically, if the SIP signalling address is
88.215.61.195, the media address will be 88.215.61.196. This is variable and depends upon which Session Border Controller your endpoint is built on.
We route media using RTP ports 6,000-40,000. You must ensure that this port range is not blocked on any firewall or router in the call path.
Session Description Protocol
Session Description Protocol enables negotiation between SIP User Agents, allowing them to agree on codec, as well as specifying the public IP address to which the receiving UA is to return media. Information relating to RTP ports is also included.
we predominately support INVITE with SDP, also known as 'Early offer'.
SIP The Session Initiation Protocol (SIP) is a signalling protocol for initiating, managing and terminating voice and video sessions across packet networks.
It is detailed in IETF RFC 3261. (http://www.ietf.org/rfc/rfc3261.txt)
The SIP protocol is an Application Layer protocol designed to be independent of the underlying Transport Layer.
Our SIP primarily utilises UDP, with the exceptions for Microsoft Lync which can be transported as TCP, without the need for a Session Border Controller to handle protocol conversion.
Short Codes The SIP trunking service supports routing to the following short codes:
999 (Access to the Emergency services)
100 (Access to Operator Assistance)
101 (The national single non-emergency number for the Police Force)
111 (The national single non-emergency number for the NHS)
112 (Access to the Emergency services)
116 xxx (Harmonised Services of Social value)
118 (UK Directory enquiries)
123 (Access to Speaking Clock)
18000* to *18009 (Access to Voice Text Services for the Deaf)
195 (Access to Blind & Disabled Directory Enquiry Facilities)
SIP
Responses
SIP responses are the codes used by Session Initiation Protocol for communication. They complement the SIP Requests, which are used to initiate action such as a phone conversation. Note that the Reason Phrases of the responses listed below are only the recommended examples, and can be replaced with local equivalents without affecting the protocol.
[http://en.wikipedia.org/wiki/List_of_SIP_response_codes]
See separate section at the end of this document for more information.
SIP
Signalling
Our SIP Trunking provides SIP signalling as a method for Communication Providers to inter-connect with our VoIP network, supporting VoIP to VoIP calling as well as calls to/from the PSTN.
Our SIP Trunking supports the transport of SIP signalling messages using UDP. SIP messages sent using TCP, TLS, SCTP and IPSEC are not supported at present. SIP (V1 and V2) are supported between network and the customer CPE. The SIP standard is documented in the Internet Engineering Task Force (IETF) RFC 3261. Please note that this service does not support H.323, SIP-I, SIP-D and SIP-T.
SIP
Standards
SIP (V1 and V2) are supported between the Gamma network and the customer CPE. The SIP standard is documented in the Internet Engineering Task Force (IETF) RFC 3261. The IETF SIP Working Group. (http://www.ietf.org/rfc/rfc3261.txt). On this page, you'll find all current Internet Drafts, RFCs and standards
SIP to ISUP Cause Codes
Below is a list of commonly-encountered ISDN cause codes with corresponding SIP responses (this list is not exhaustive):
01 - Unallocated number - 404 not found
17 - User busy - 486 busy here
19 - No answer from user - 480 temporarily unavailable
22 - Number changed - 410 gone
28 - Address incomplete - 484 address incomplete
38 - Network out of order - 503 service unavailable
SIP to ISUP inter-networking is defined in RFC3398.
SMOKEPING Smokeping is a monitoring platform from which we are able to poll our customer's public IP address. This enables the determination of jitter, packet loss and latency, in instances of reported call quality problems.
TRACING 'Tracing' refers to the running of Wireshark captures, which facilitate the analysis of SIP requests and responses between User Agents. We run continual signalling traces on the majority of its SIP endpoints, but do not store media historically due to attendant capacity limitations. Media traces are initiated on an adhoc basis, for pre-determined periods.
User Agent SIP User-Agent is an endpoint entity which initiates and terminates sessions by exchanging requests and responses
The User agents consists of two components:
User Agent Server UAS (replay for requests)
VIA HEADER The VIA header in a SIP INVITE defines the transport protocol, port number and public
IP address of the originating User Agent. Whilst the initial INVITE contains only one VIA record, this field is used to track the path of the request, and helps to prevent looping. If the VIA header contains a private IP address, our signalling gateway will issue a 403 forbidden SIP response
VOIP VOIP is an acronym for Voice Over Internet Protocol, or in more common terms phone
service over the Internet.
If you have a reasonable quality Internet connection you can get phone service delivered through your Internet connection instead of from your local phone company.
SIP Response Description
ACK Confirms that the client has received a final response to an INVITE request.
BYE Terminates a call and can be sent by either the caller or the callee.
CANCEL Cancels any pending request.
INVITE Indicates a client is being invited to participate in a call session.
MESSAGE Transports instant messages using SIP.
NOTIFY Notify the subscriber of a new Event.
OPTIONS Your endpoint may send our SBC address OPTIONS messages at intervals, as a
keep-alive.
PRACK Provisional acknowledgement. If your SIP INVITE supports 100 rel, we will expect to
receive a PRACK
REGISTER Register messages enable us to route calls to an IP address; only pertinent to
SIP-registrar services.
SUBSCRIBE Subscribes for an Event of Notification from the Notifier.
