Session Description Protocol

When a call is initiated between two endpoints it is important to have information of calling and called party IP address both for signaling and media, codec they are going to use and several other important information. In a SIP based call (call where SIP protocol is used) we use SDP to advertise media information like transport address, transport protocol, codec, ports and other session description information. SDP works at application layer.  SDP is defined in RFC 2327, 4566 (newer). Normally we see SDP in Invite, 200 OK or ACK depending on the call type which we will discuss soon.

SIP Response Codes

In this post I am going to discuss about SIP Response codes. In the previous post we discussed about SIP messages. SIP Response codes as the name says is a kind of response to any SIP message. It can be a response to Invite message, Bye message or an Options message. We have. 6 different classes SIP Response codes which we will discuss now.

1XX - INFORMATIONAL RESPONSE

2XX - SUCCESSFUL RESPONSES

3XX - REDIRECTION RESPONSES

4XX - CLIENT FAILURE RESPONSES

5XX - SERVER FAILURE RESPONSES

6XX - GLOBAL FAILURE RESPONSES

SIP Messages

When we talk about SIP call flow there are two important things that need to be understood SIP messages and response code. SIP messages and responses are used for setup , in session changes, info and for  tearing down a session. A response code is a response which is sent in response to a SIP message, and we can also get a SIP message in acknowledgement of response code.We will look into some quite frequent SIP messages in this post and response codes in the next post.

SIP Messages:

INVITE: Invite is very first message to set up a connection. It is also used to modify a session as re-invite (which we will discuss in later posts). User Agent sending an Invite is known as User Agent Client.

ACK: ACK is sent in response to a 200 OK message. ACK is used only with Invite or a Re-Invite message.

SIP CALL FLOW

In SIP Basics we discussed about components of SIP. It's time to understand a sample call flow. Before we start I would like to discuss few components of call flow:

Request: When a UAC (User Agent Client) wants to initiate a call it sends an Invite to the UAS (User Agent Server). In this case Invite is a request .

Response: A reply to UAC's request is called a response. In this case it would be 100 Trying, which we will discuss in a moment.

SIP BASICS

SIP BASICS

SIP:

SIP Stands for Session Initiation Protocol. SIP is becoming a global standard for VoIP. SIP is a protocol which is used for signalling, controlling multimedia sessions such as voice and video between two endpoints or through a proxy server. SIP is also used for instant messaging, client presence, user mobility based on its current location.

  

SIP has been developed by IETF: Internet Engineering Task Force. This group is responsible for adding new features and modifying existing features in SIP. All concepts used in SIP are defined under RFC’s i.e. RFC 3261 which is one of the most important RFC for VoIP.

Analog and Digital Voice System

Analog Systems:

   When we thing of voice systems the very first thing that comes into our mind is a home phone. Like our home phones our old telephony systems used to operate in analog mode. It has two cables Ring and Tip and it operates on -48 DC voltage. When the circuit between the two gets complete it withdraws a dial tone from the central office. The voice waveform of these kind of calls are analog waveform.

MEDIA RESOURCES

A media resource is a software-based or hardware-based entity that performs media processing functions on the data streams to which it is connected. Media processing functions include mixing multiple streams to create one output stream (conferencing), passing the stream from one connection to another (media termination point), converting the data stream from one compression type to another (transcoding), streaming music to callers on hold (music on hold), echo cancellation, signaling, voice termination from a TDM circuit (coding/decoding), packetization of a stream, streaming audio (annunciation), and so forth. The software-based resources are provided by the Cisco Unified Communications Manager (Unified CM) IP Voice Media Streaming Service (IP VMS). Digital signal processor (DSP) cards provide both software and hardware based resources.

Media Termination Point

MTP

A media termination point (MTP) is an entity that accepts two full-duplex media streams. It bridges the streams together and allows them to be set up and torn down independently. The streaming data received

from the input stream on one connection is passed to the output stream on the other connection, and vice versa.

CSS Line Device Approach: Path Selection

CSS Line Device Approach: Path Selection

Hi All,

  In the previous post I explained how Line Device CSS approach works, but I also mentioned:

• ·        The Device CSS decides through which gateway/trunk call will go out of our network.
• ·        Line CSS defines (Class of Service) calling privileges.

In this post we will try to understand how device CSS helps us to select the right gateway. Although Cisco has introduced a concept of local gateway, but we will discuss that in a different post.

CSS Line Device Approach

CSS Line Device Approach

Hi All,

  As we start understanding CSS and partitions next confusion we normally have is what is Line and Device CSS. There is a very simple definition of CSS Line/Device Approach

• ·        The Device CSS decides through which gateway/trunk call will go out of our network.
• ·        Line CSS defines (Class of Service) calling privileges.

CSS & Partition

 

Hi All,

  The most basic and complicated topic in CUCM dial-plan is CSS &Partition. When I look at this topic I compare it with Lock (partition) and Key (CSS) and Route Pattern or a DN as a door. Assume CSS as a key chain with different keys as partition.

Integrating VMware with GNS3

Hi All,

  This is one of the toughest situation I faced when I tried to integrate VMware and GNS3 as I had to simulate a voice gateway working with CUCM and as I said its a no hardware lab (excludes my PC). After 2 months of trails and failures finally I found a way to do it.

Managing VMware in Computer

Hi All,

   There was a big problem with me when I started with VMware that how to configure my VMware so I can access my VMware application like Cisco Call Manager, Cisco Unity Connection etc .

Installing CUCM on VMware

Hi Friends I found this Video useful for installing CUCM on VMware.

  If there is any query ask me .

Configuring Call Manager Express in GNS3

Connecting your computer with GNS3

Its very important that our computer communicates with GNS3 in able to visualize the things in the real scenario. In this post we will go stet by step and check how we can make this happen.

Getting Familiar with GNS3

 People who are not aware of GNS3 , just to give a brief about GNS3.

GNS3 is a graphical network simulator that allows simulation of complex networks.

GNS3 is an excellent complementary tool to real labs for network engineers, administrators and people wanting to study for certifications such as Cisco CCNA, CCNP, CCIP and CCIE as well as Juniper JNCIA, JNCIS and JNCIE.

  

About this blog

I love to learn about the Cisco voice technologies, but the problem most of us face is that we dont have real hardware and platforms to work upon. Then the last option we are left with is virtual lab. I started my first lab with GNS3 where I made my first IP phone work. That day I was very happy but that took 3 months for me . I did everything correct but RAM was less on the router.
   Similarly while preparing for CCNA voice and later we face lots of issues because we don't get much help how to configure different things and we get stuck on simple issues.