VoIP protocols, part 1. There are many other protocols involved in IP telephony that function alongside SIP or even in the place of it. It is important for a telecom engineer to be familiar with these protocols, to know what they do and how they can be leveraged on the telecommunications network. In a future article, we will focus on protocols that can be used as alternatives to SIP.
SIP is a protocol used by both voice and video endpoints to provide call setup, signaling, and call teardown for communication sessions. All of these endpoints register to a SIP server, also known as an IP PBX, which is used to coordinate advanced features such as call transfer, call hold, music on hold, and other traditional and enhanced telephony features. To enable even more specialized services, such as call routing, call queuing, Interactive Voice Response, multiparty voice and video conferencing, integration with web services, and interconnectivity with the traditional PSTN, additional servers and devices can be added to a VoIP network.
Keep in mind that SIP does not carry the voice or video itself. Instead, it operates in conjunction with several other protocols that carry the session media.
Companion protocols that work in conjunction with SIP. The following diagram shows the various protocols that are used in conjunction with SIP in a typical VoIP telephony conversation. These protocols are mapped to their relative layers within the OSI model framework. This means that SIP is independent of the protocols used at lower layers, so it can work with any Transport Layer protocol. Specifically, it enables endpoints to negotiate the media type, format, and all of its associated properties.
SDP does not carry the media itself, nor is it sent via any Transport Layer protocol. The mechanisms for the associated profile and payload format, referenced in the design of the RTP architecture , are implemented on the level of the application layer, instead of the operating system layer.
Applications such as VoIP that need to employ real-time streaming of multimedia data, typically require the timely delivery of data, with varying tolerance in packet loss. As an example, audio packet loss in a VoIP application can cause losing some milliseconds of audio data. This loss can be appropriately handled by error compensation algorithms to make it insignificant and imperceptible to the caller s. TCP Transmission Control Protocol is also standardized for RTP use, even though it is not typically employed in applications due to its error-control mechanisms that can cause delays and affect timely packet delivery.
Check your inbox! Click on the button in the email body to verify your email address - if you can not find it, check your spam folder. The 1 Communications System! Sitemap Privacy. Get it Free for 1 Year Pricing. What are the advantages and usage of RTP?
The source can exploit this information that carries in real-time to adapt the type of coding to the level of available resources. It also ensures the communication of information, identification of participants of an RTP session.
Thus everyone can know how many participants are part of the conference. Finally, it adjusts the transmission rate. This information will be used to improve the output rate and adapt it to accommodate all people wishing to join the event.
The confidentiality of media flows is achieved by encryption. As the data compression used with the payload formats described in this profile is applied end-to-end, the encryption can be done after the compression so that there is no conflict between the two operations. A potential denial of service threat exists for data encodings that use compression techniques with a computing load different from that of the receiving end.
Under extreme conditions, an attacking potential can inject into the flow of complex pathogenic datagrams to be decoded that cause the overload of the recipient.
As with any IP-based protocol, a receiver may be overloaded merely by receiving too many desired or unwanted packets in some circumstances. It may have audio conversion options in addition to all of its Capture Utility and Frame Analyzer capabilities.
For example, it would be possible to know how a hacker with access to a LAN can easily sniff it, analyze RTP streams and listen to a particular conversation.
This type of software exists for use in a Windows or Linux environment. In both cases, a graphical interface facilitates the commands of such software for backup and exploitation. Sign In. Reza Mousavi. OSI Model RTP can be described as a UDP add-in that adds to each transmitted packet valuable information about the sequence number which will put the received packets back in order plus a packet timestamp for the database restore.
From a technical point of view, RTP allows : — Reconstitute the time base of the audio, video, and real-time data streams in general.
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