bilal ghayyad
2007-Jul-03 22:06 UTC
[asterisk-users] Determining the used codec for the IP Trunk (SIP Trunk)
Hi List; Where I determine the codec to be used for the SIP Trunk (between Asterik and another SIP softswitch)? Regards Bilal ____________________________________________________________________________________ Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545433
Mojo with Horan & Company, LLC
2007-Jul-03 22:27 UTC
[asterisk-users] Determining the used codec for the IP Trunk (SIP Trunk)
you might use sip show peer peername to see what a peer will allow or show channel channelname (channel name as retrieved from show channels) to determine what a current conversation is using Moj bilal ghayyad wrote:> Hi List; > > Where I determine the codec to be used for the SIP > Trunk (between Asterik and another SIP softswitch)? > > Regards > Bilal > > > > ____________________________________________________________________________________ > Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. > http://answers.yahoo.com/dir/?link=list&sid=396545433 > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Alex Balashov
2007-Jul-03 22:31 UTC
[asterisk-users] Determining the used codec for the IP Trunk (SIP Trunk)
On Tue, 3 Jul 2007, bilal ghayyad wrote:> Where I determine the codec to be used for the SIP Trunk (between > Asterik and another SIP softswitch)?Are you asking positively how to determine which codec is being negotiated between those two elements, or, normatively, which one is best to use? If the former question, you can look at the SDP (Session Description Protocol) payload in the INVITEs (and other messages part of the INVITE transaction). Find the 'rtpmap' lines. They look like this: a=rtpmap:18 G729/8000 There may be multiple such lines indicating that endpoint's support for all of them. If so, the only way to determine which one is actually going to be used is to look at the RTP stream with a protocol analyser. It should be able to tell you. Wireshark/Ethereal certainly can. If you're asking which codec should be used, it depends on the desired application, whether the trunk is running over a LAN, WAN, or over the public Internet, whether QoS is involved, etc. -- Alex -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671
bilal ghayyad
2007-Jul-05 21:50 UTC
[asterisk-users] Determining the used codec for the IP Trunk (SIP Trunk)
Dear Alex; I am asking about: What is the configuration that I can do it to let the traffic between the two Asterisk PBX and another IP BX to be g729 or G711 or g723? In other words, how can I let the ued codec for the IP Trunk between my Asterisk and the other IP PBX to be g729 and not g711? Ofcourse, I am assuming that the other side also supporting g729. Regards Bilal> Where I determine the codec to be used for the SIPTrunk (between> Asterik and another SIP softswitch)?Are you asking positively how to determine which codec is being negotiated between those two elements, or, normatively, which one is best to use? If the former question, you can look at the SDP (Session Description Protocol) payload in the INVITEs (and other messages part of the INVITE transaction). Find the 'rtpmap' lines. They look like this: a=rtpmap:18 G729/8000 There may be multiple such lines indicating that endpoint's support for all of them. If so, the only way to determine which one is actually going to be used is to look at the RTP stream with a protocol analyser. It should be able to tell you. Wireshark/Ethereal certainly can. If you're asking which codec should be used, it depends on the desired application, whether the trunk is running over a LAN, WAN, or over the public Internet, whether QoS is involved, etc. ____________________________________________________________________________________ It's here! Your new message! Get new email alerts with the free Yahoo! Toolbar. http://tools.search.yahoo.com/toolbar/features/mail/
Alex Balashov
2007-Jul-05 22:01 UTC
[asterisk-users] Determining the used codec for the IP Trunk (SIP Trunk)
Bilal, There are allow= options you can use on the peers in sip.conf to define what codec capability Asterisk advertises toward them, and therefore, what the negotiation on both call legs will ultimately settle upon. When those legs are bridged -- as Asterisk does unto them -- they will be of a codec that is within that bounded set. You can have multiple allow lines, i.e. allow=g729 allow=ulaw ... or just one. Does that help? -- Alex -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671
Noah Miller
2007-Jul-05 23:54 UTC
[asterisk-users] Determining the used codec for the IP Trunk (SIP Trunk)
Hi Bilal ->> In other words, how can I let the ued codec for the IP >> Trunk between my Asterisk and the other IP PBX to be >> g729 and not g711? Ofcourse, I am assuming that the >> other side also supporting g729. >> > You can have multiple allow lines, i.e. > > allow=g729 > allow=ulawBe sure to also put in a disallow statement first: disallow=all allow=g729 - Noah