- This topic has 4 replies, 1 voice, and was last updated 16 years, 10 months ago by John Mac Duf.
28th May 2004 at 08:54 #27248John Mac DufGuest
I intend to use an IP network to carry the signaling between some Alcatel Omni PCX 4400. Apparently this can be done with no problem …but I can’t find any recommendation about the amount of IP traffic required to support this signalling !?
I would be interested to know how much IP traffic I should consider for each voice channel. Any help would be much appreciated !!
Thanks in advance,28th May 2004 at 16:40 #27249Tony FGuest
I’m not sure if when you say signalling you use the term ‘signalling’ to mean the entire call or if you actually mean the signalling used to setup and clear down calls. I’ll assume that you mean the signalling to setup and clear down calls.
With regards to UDP based VoIP signalling i.e. SIP, unless you are planning to route your signalling over a separate IP network/path then you do not need to know how much bandwidth is used for signalling. The reason being that the voice CODEC will always use (even based upon todays low bandwidth CODEC’s) more bandwidth than the signalling. As you will not be signalling at the same time as voice is being transmitted if you have sufficient bandwidth for the voice you will have sufficent bandwidth for the signalling.
There are certain rare scenarios which create exceptions to this. However, for almost all existing deployments and requirements, these do not apply. These exceptions are:
1) If the signalling is used to
2) If the signalling is constantly
in use (i.e. T.120).
3) If large number of
supplementary services are used
during the call.
For basic voice the CODECs will use more bandwidth than the signalling.
With regards to TCP based signalling – usually H.323 but H.323 can use UDP for call signalling (and SIP can use TCP) most of the above still applies but you will also have the overhead of the TCP keepalive packets (for H.323 you may have two lots of TCP packets – those for H.225.0 and those for H.245).
With regards to the bandwidth used for each voice channel it will depend upon the CODEC used, CODEC frames per IP packet and if voice activity detection (VAD)is used.
Do you know which CODEC you plan on using? If not what level of quality are you looking for?
Tony30th May 2004 at 21:54 #27250Alaa MusaGuest
I have a VoIP gateway connected to ZTE PBX , In PSTN they are using R2 signalling for E1s , the problem I have that some numbers o can’t reach them from E1s it give me switch congestion , but if i used a normal phone line with the same PSTN i can reach these numbers , couls any one help me to know where is the problem.4th June 2004 at 11:03 #27251mohsinGuest
I interested in your explanation regarding signalling over IP. Now everybody talk about CCS7 over IP or better known as SIGTRAN. My question is, hom many message bit rate per sub in SIGTRAN inveronment.7th June 2004 at 08:57 #27252John Mac DufGuest
Thanks all for your answers !
Well, the thing is that I actually want to carry the signalling over a separate IP network, and leave (for the time being) the voice traffic outside this IP network.
So basically, I’m just trying to know how much bandwidth is required on this IP network to carry the signaling traffic (=set up and clear down the calls), knowing that the Alcatel equipments use a proprietary signaling protocol (called ABC-F) which seems to be a variant of Q-Sig, but on which I can’t hardly find any information !?
Again, thanks in advance for any help you can give me here !