RE: [Asterisk-Users] SIP audio port usage
It depends on the ATA, and our router, etc... Typically in the range between 1 and 2 --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of -Adrien Laurent -Sent: Monday, September 19, 2005 12:23 PM -To: asterisk-users@lists.digium.com -Subject: [Asterisk-Users] SIP audio port usage - -Hi, - -I know that SIP is using port 5060 for session initiation, -but which port does it use for audio ? is it dynamically assigned ? - -Thanks, - -Adrien - --- -Adrien Laurent - CIO -www.modulis.ca -514-284-2020 ext 202 -[EMAIL PROTECTED] - - -___ ---Bandwidth and Colocation sponsored by Easynews.com -- - -Asterisk-Users mailing list -Asterisk-Users@lists.digium.com -http://lists.digium.com/mailman/listinfo/asterisk-users -To UNSUBSCRIBE or update options visit: - http://lists.digium.com/mailman/listinfo/asterisk-users - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP audio port usage
So the more reliable way to do QoS is with MAC adress and not on a port basis. Am I right ? Thanks for your help, Adrien On 9/19/05, Rich Adamson [EMAIL PROTECTED] wrote: I know that SIP is using port 5060 for session initiation, but which port does it use for audio ? is it dynamically assigned ? Its dynamically assigned on a per-call basis. Asterisk assigns the port based on contents of rtp.conf. Remote sip phones assign port numbers based on whatever the manufacturer happened to choose (no industry standard). E.g., Cisco uses 32,768 to something around 40,000, while xlite uses something in the area of 8,000. The various manufacturers are not consistent at all. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Adrien Laurent [EMAIL PROTECTED] www.modulis.ca ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP audio port usage
Yes, because then the MACs specified would be getting the QoS, not just certain ports. This is how I set up my customers when they have QoS available. --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of -Adrien Laurent -Sent: Tuesday, September 20, 2005 8:53 AM -To: Asterisk Users Mailing List - Non-Commercial Discussion -Subject: Re: [Asterisk-Users] SIP audio port usage - -So the more reliable way to do QoS is with MAC adress and not -on a port basis. -Am I right ? - -Thanks for your help, - -Adrien - -On 9/19/05, Rich Adamson [EMAIL PROTECTED] wrote: - - I know that SIP is using port 5060 for session -initiation, but which - port does it use for audio ? is it dynamically assigned ? - - Its dynamically assigned on a per-call basis. - - Asterisk assigns the port based on contents of rtp.conf. - - Remote sip phones assign port numbers based on whatever the - manufacturer happened to choose (no industry standard). E.g., Cisco - uses 32,768 to something around 40,000, while xlite uses -something in the area of 8,000. - The various manufacturers are not consistent at all. - - - - ___ - --Bandwidth and Colocation sponsored by Easynews.com -- - - Asterisk-Users mailing list - Asterisk-Users@lists.digium.com - http://lists.digium.com/mailman/listinfo/asterisk-users - To UNSUBSCRIBE or update options visit: -http://lists.digium.com/mailman/listinfo/asterisk-users - - - --- -Adrien Laurent -[EMAIL PROTECTED] -www.modulis.ca -___ ---Bandwidth and Colocation sponsored by Easynews.com -- - -Asterisk-Users mailing list -Asterisk-Users@lists.digium.com -http://lists.digium.com/mailman/listinfo/asterisk-users -To UNSUBSCRIBE or update options visit: - http://lists.digium.com/mailman/listinfo/asterisk-users - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP audio port usage
But, if I have Xlite running on client PC and at the same time the user is doing FTP, both service has the same QoS treatment? Is there a way to differentiate these services besides the port? Sebastian On 9/20/05, Sherwood McGowan [EMAIL PROTECTED] wrote: Yes, because then the MACs specified would be getting the QoS, not just certain ports. This is how I set up my customers when they have QoS available. --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of -Adrien Laurent -Sent: Tuesday, September 20, 2005 8:53 AM -To: Asterisk Users Mailing List - Non-Commercial Discussion -Subject: Re: [Asterisk-Users] SIP audio port usage - -So the more reliable way to do QoS is with MAC adress and not -on a port basis. -Am I right ? - -Thanks for your help, - -Adrien - -On 9/19/05, Rich Adamson [EMAIL PROTECTED] wrote: - - I know that SIP is using port 5060 for session -initiation, but which - port does it use for audio ? is it dynamically assigned ? - - Its dynamically assigned on a per-call basis. - - Asterisk assigns the port based on contents of rtp.conf. - - Remote sip phones assign port numbers based on whatever the - manufacturer happened to choose (no industry standard). E.g., Cisco - uses 32,768 to something around 40,000, while xlite uses -something in the area of 8,000. - The various manufacturers are not consistent at all. - - - - ___ - --Bandwidth and Colocation sponsored by Easynews.com -- - - Asterisk-Users mailing list - Asterisk-Users@lists.digium.com - http://lists.digium.com/mailman/listinfo/asterisk-users - To UNSUBSCRIBE or update options visit: -http://lists.digium.com/mailman/listinfo/asterisk-users - - - --- -Adrien Laurent -[EMAIL PROTECTED] -www.modulis.ca -___ ---Bandwidth and Colocation sponsored by Easynews.com -- - -Asterisk-Users mailing list -Asterisk-Users@lists.digium.com -http://lists.digium.com/mailman/listinfo/asterisk-users -To UNSUBSCRIBE or update options visit: - http://lists.digium.com/mailman/listinfo/asterisk-users - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP audio port usage
Then you'll have to make sure that other services are lower QoS. Past that, find out what port XLITE uses and then QoS that port. --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of -Sebastian Milioto -Sent: Tuesday, September 20, 2005 9:50 AM -To: Asterisk Users Mailing List - Non-Commercial Discussion -Subject: Re: [Asterisk-Users] SIP audio port usage - -But, if I have Xlite running on client PC and at the same -time the user is doing FTP, both service has the same QoS treatment? -Is there a way to differentiate these services besides the port? - -Sebastian - - - -On 9/20/05, Sherwood McGowan [EMAIL PROTECTED] wrote: - Yes, because then the MACs specified would be getting the QoS, not - just certain ports. This is how I set up my customers when -they have - QoS available. - - --Original Message- - -From: [EMAIL PROTECTED] - -[mailto:[EMAIL PROTECTED] On -Behalf Of Adrien - -Laurent - -Sent: Tuesday, September 20, 2005 8:53 AM - -To: Asterisk Users Mailing List - Non-Commercial Discussion - -Subject: Re: [Asterisk-Users] SIP audio port usage - - - -So the more reliable way to do QoS is with MAC adress and -not on a - -port basis. - -Am I right ? - - - -Thanks for your help, - - - -Adrien - - - -On 9/19/05, Rich Adamson [EMAIL PROTECTED] wrote: - - - - I know that SIP is using port 5060 for session - -initiation, but which - - port does it use for audio ? is it dynamically assigned ? - - - - Its dynamically assigned on a per-call basis. - - - - Asterisk assigns the port based on contents of rtp.conf. - - - - Remote sip phones assign port numbers based on whatever the - - manufacturer happened to choose (no industry standard). E.g., - - Cisco uses 32,768 to something around 40,000, while xlite uses - -something in the area of 8,000. - - The various manufacturers are not consistent at all. - - - - - - - - ___ - - --Bandwidth and Colocation sponsored by Easynews.com -- - - - - Asterisk-Users mailing list - - Asterisk-Users@lists.digium.com - - http://lists.digium.com/mailman/listinfo/asterisk-users - - To UNSUBSCRIBE or update options visit: - -http://lists.digium.com/mailman/listinfo/asterisk-users - - - - - - - --- - -Adrien Laurent - -[EMAIL PROTECTED] - -www.modulis.ca - -___ - ---Bandwidth and Colocation sponsored by Easynews.com -- - - - -Asterisk-Users mailing list - -Asterisk-Users@lists.digium.com - -http://lists.digium.com/mailman/listinfo/asterisk-users - -To UNSUBSCRIBE or update options visit: - - http://lists.digium.com/mailman/listinfo/asterisk-users - - - - - ___ - --Bandwidth and Colocation sponsored by Easynews.com -- - - Asterisk-Users mailing list - Asterisk-Users@lists.digium.com - http://lists.digium.com/mailman/listinfo/asterisk-users - To UNSUBSCRIBE or update options visit: -http://lists.digium.com/mailman/listinfo/asterisk-users - -___ ---Bandwidth and Colocation sponsored by Easynews.com -- - -Asterisk-Users mailing list -Asterisk-Users@lists.digium.com -http://lists.digium.com/mailman/listinfo/asterisk-users -To UNSUBSCRIBE or update options visit: - http://lists.digium.com/mailman/listinfo/asterisk-users - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP audio port usage
So the more reliable way to do QoS is with MAC adress and not on a port basis. Am I right ? Thanks for your help, Adrien On 9/19/05, Rich Adamson [EMAIL PROTECTED] wrote: I know that SIP is using port 5060 for session initiation, but which port does it use for audio ? is it dynamically assigned ? Its dynamically assigned on a per-call basis. Asterisk assigns the port based on contents of rtp.conf. Remote sip phones assign port numbers based on whatever the manufacturer happened to choose (no industry standard). E.g., Cisco uses 32,768 to something around 40,000, while xlite uses something in the area of 8,000. The various manufacturers are not consistent at all. A very common way of handling QoS is to rely on the TOS (Type of Service) bits located in the IP header. Those bits are set in asterisk packets via a statement like: tos=lowdelay in sip.conf and iax.conf. There are similar type parameters available in most quality sip phones. However, once the bits are set properly, its then up to your router and/or switch to queue the packets properly for transmission over the network. The majority of the soho routers and switches do not have code to actually handle that queuing, and even if you have a device that does properly handle it, the prioritization of the packets is outbound traffic only. Your internet service provider would have to do something to prioritize the inbound traffic to you, and most won't do that. In addition, the majority of the backbone Internet providers don't pay any attention to any QoS settings. The QoS parameters work very nicely in corporate networks where support personnel understand the concepts and monitor their resources, but isp's and itsp's generally don't have a clue (or don't care). There are other software packages that will help prioritize packets to/from the Internet, and most of them use some form of trickery to accomplish the goal. For example, outbound http packets are delayed allowing rtp packets to be sent without delay, resulting in a form of QoS. By delaying the http packets (sent outbound), the remote web server essentially is placed in a wait state causing it not to forward any packets to your site, resulting in a form of inbound QoS. Those types of QoS will not handle streaming packets such as those associated with listening to music or watching videos. For the most part, QoS across the Internet (regardless of whose equipment you use) is not very effective today since the majority of isp's and backbone suppliers have not implemented QoS. As one example, you could have the most expensive Cisco router on your end with properly implemented QoS prioritization, but if I sent a large number of icmp or other fake packets to your IP address, I'd consume all available bandwidth leaving your rtp packets no way to reach your site reliably. For home and small offices that rely on DSL type facilities, implementing QoS can improve the quality as generally the outbound bandwidth is significantly less then inbound bandwidth. In those cases, prioritizing outbound traffic (on the low bandwidth portion) may help, but it still won't do much for inbound traffic. The exact same issues apply regardless of whether you rely on the TOS bits or MAC address method of prioritizing traffic. The TOS bits just happens to be a far more common method. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP audio port usage
I know that SIP is using port 5060 for session initiation, but which port does it use for audio ? is it dynamically assigned ? Its dynamically assigned on a per-call basis. Asterisk assigns the port based on contents of rtp.conf. Remote sip phones assign port numbers based on whatever the manufacturer happened to choose (no industry standard). E.g., Cisco uses 32,768 to something around 40,000, while xlite uses something in the area of 8,000. The various manufacturers are not consistent at all. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
