Cisco Collaboration: Cisco CallManager Express (CME) SIP Trunking
Starting from the IP telephony systems, IP PBXs are starting to predominate in the business of the Voice technology, and the TDM PBXs are no longer the primary source as the crossover going between two Voice networks. The usage of the TDM PBXs has decreased in the last couple of years, and the use of the IP PBX is becoming a good investment in IP LANs and WANs.
In order to connect to the PSTN, PBXs need some sort of trunking such as TDM (T1/E1) or analog lines. IP PBXs can access the PSTN using these types of trunks, but need a media gateway that converts the IP voice traffic to traditional PSTN, which sometimes can result in successive translation from IP domain to TDM domain. These successivetranslations increase the maintenance costs of the gateways, increases latency, and reduces voice quality.
SIP is an ASCII based, application-layer control protocol that can be used to establish, maintain, and terminate calls between two or more endpoints. SIP has rapidly emerged as the standard protocol used in IP communications, because it is a multimedia protocol that can be used for video sessions and instant messaging in addition to voice. Also, SIP can handle conference sessions and broadcasts, as well as one-to-one sessions. SIP has great potential in transforming and developing the way people communicate. For this reason, Cisco has and continues to play an important role in taking a leadership to create new technologies that make SIP and its applications the standard of IP communications.
SIP trunks are similar to a phone line, except that SIP trunks use the IP network, not the PSTN. In addition, SIP trunks permit the convergence of voice and data onto common all-IP connections. In order to access the IP network using a SIP trunk, it is necessary that configurations be made on the service provider, as well as on the customer side.
Customers need to set and configure CME, which is the PBX that will interpret the SIP signal adequately and pass traffic successfully. The service provider needs to configure an SIP Proxy Server. However, SIP trunks are more complicated to establish than regular PSTN trunks. The reason is that a customer faces challenges in handling different interpretation and implementations of SIP by equipment vendors, delivering security, managing quality of service (QoS), enabling Network Address Translation (NAT) and firewall traversal, and ensuring carrier-grade reliability and continuity of service.
Cisco CME is an IP telephony solution that is integrated directly into Cisco IOS software. CME permits small and medium businesses to deploy voice, data, and video on a single platform. An IP telephony network is simple to set because CME runs on a single router, which delivers a PBX functionality for businesses. Therefore, by using CME, small and medium businesses can deliver IP telephony and data routing using a single converged solution with minimal costs.
CME started to support SIP trunking when CME 3.1 was released. However, some problems existed when an SIP phone called an SCCP phone or tried to access voicemail. The problem is that SCCP phones connected to CME require the use of out-of-band DTMF relay to transport DTMF (digits) across VoIP connections, and SIP phones use in-band transports. A DTMF distortion existed between the two devices. When CME 3.2 was released, support was added to the DTMF relay. DTMF digits from SCCP could be converted to in-band DTMF relay mechanism through RFC2833 or Notify methods.
CME currently supports this list of DTMF internetworking for SIP to SIP calls:
Notify <—> Notify since 12.4(4)T
RFC2833 <—> Notify since 12.4(4)T
Notify <—> RFC2833 since 12.4(4)T
Inband G711 <—> since 12.4(11)T [Requires Transcoder]
Codec Support and Transcoding
Another important aspect to consider when you set up an SIP trunk is the codecs supported. Codecs represent the pulse-code modulation sample for signals in voice frequencies. SIP trunks support these codecs: G.711 and G.729. However, for different features such as Cisco Unity Express (CUE) and Music on Hold (MOH), only codec G.711 is supported.
This means that voice calls that use SIP trunks using codec G.729 cannot access CUE, unless a transcoder exists to permit the compression and decompression of voice streams to match the CUE capabilities. MOH can also use codec G.729 to save bandwidth, but the codec does not provide adequate quality MOH streams. This is due to the fact that G.729 is optimized for speech. Therefore, you must force MOH to use G.711.
When a call comes in on a SIP trunk and gets forwarded (CFNA / CFB / CFA), then the default behavior is for the CME to send the 302 “Moved Temporarily” SIP message to the Service Provider (SP) proxy. The user portion of the Contact Header in the 302 message might need to be translated to reflect a DID that the SP proxy can route to. The host portion of the Contact Header in the 302 message should be modified to reflect the Address of Record (AOR) using the host-registrar CLI under sip-ua and the b2bua CLI under the VoIP dial peer going to the CUE.
Some SIP proxies might not support this. If so, then you need to add this:
RouteXP_router(config)#voice service voip
RouteXP_router(conf-voi-serv)#no supplementary-service sip moved-temporarily
When a call comes in on an SIP trunk to an SCCP Phone or CUE AutoAttendant (AA) and is transferred, the CME by default will send a SIP REFER message to the SP proxy. Most SP Proxy Servers do not support the REFER method. This needs to be configured in order to force the CME to hairpin the call:
RouteXP_router(config)#voice service voip
RouteXP_router(conf-voi-serv)#no supplementary-service sip refer
If an SCCP phone places a call from PSTN on HOLD, the CME locally changes the media. No SIP messages are sent across on the SIP trunk. Music on Hold will be played to the user across the SIP trunk based on the CME configuration.