That's not planned. IAX2 is really different from SIP. I'll try first to make
something suitable in SIP :).
As far as I know, pjsip, the native sip stack on which csipsimple is based, doesn't
support IAX2.
If the pjsip team decide to add something for IAX2, csipsimple will have this
improvement too.
Another solution would be to integrate another native opensource and portable stack,
but that's really far to be a priority.
I don't reject this request, but unless somebody else wants to contribute for that,
it is postponed for a long time.
I vote for this because of its smaller bandwidth requirements and better reliability
through firewalls. (Stun not required.) Huge advantage for android users on limited
data plans.
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as soon as I opened port 5060 to the world I got several dict attacks on sip login which basically DOS'ed my asterisk server, I had to firewall 5060 to just a few known IP's that will use my 'roid phone over wifi (will trial fail2ban to see if that alleviates the attacks)
I use an IAX client on my MacBook (Loudhush or JackenIAX) and have never seen an attack on 4569 on my asterisk server
also as mentioned IAX2 uses less b/w and avoids NAT issues
apart from this THANKS FOR CSipSimple
WORKS LOVEY ON ZTE BLADE (orange san francisco) under Gingerbread 2.3.3 Cyanogenmod7 (both RC1 and nightlies 1-15 so far)
audio routed correctly to earpiece and then loudspeaker when loudspeaking is chosen
sometimes audio quality lacks despite using codec 722 - trying KEEP AWAKE DURING CALL to see if that makes a difference
keep up the great work
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IAX is a hack, a toy, a pseudo-protocol (instead of separating signalling and media as *any* modern VoIP protocol it forces both streams to go together).
IAX does not allow IM neither presence. IAX has several design issues and critical vulnerabilities and bugs in its main implementation (Asterisk).
IAX was created by Mark Spencer because he does not properly understand SIP protocol (given the old-fashion-PBX implemented in Asterisk that is not a surprise for me).
There is enough mechanisms in SIP for handling NAT, in client and server side. In the other side IAX requires that the media goes always to the central server which could be so far from both callers. This is a pain and un feasible in many environments, generating latency and so.
IAX is just supported by a few (very few) phones and just some PBX's (the anti-cool-DTMF-based Asterisk and FreeSwitch). Must we asume that CSipSimple is designed to "work with Asterisk"? Please not.
Please forget IAX. Just people with no knowledge of SIP is in favour of IAX (just because "it seems easier").
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Whether you're for or against IAX2, matters not. The fact that SIP and IAX exsist, is sufficient enough to make them accessable to any user, anywhere.
CSipSimple is a great, stable program and if IAX can be added to it, then I say go for it.
Will save me running 2 apps and if it can all function in CSipSimple, that would only be a good thing..
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Yep, actually I did a big effort on separating the SIP stack from the rest of the program. One of the next step will be to allow other sip stacks than pjsip (for example the stock gingerbread one or doubango, which is another great sip stack).
If this can be accomplished, I guess that it could be a good starting point to add other VoIP protocols support. The idea is to allow this kind of third party voip backend to be loaded as plugins apps, so would be a good compromise.
Android os is very well designed to allow to extends apps features with other plugins apps.
Well, as for this point, still open, still not in my higher priorities but contributions are welcome ;).
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@Comment 7:
IAX2 does support messaging and IIRC also presence. Combining the signaling and the audiostream into one connection has advantages in regards to protocol overhead. Additionally, an encrypted IAX2 stream is significantly harder to dissect than SIP because of the combination of signaling and audio.
One of the biggest disadvantages of SIP is that it's very hard to operate a nat'ed PBX and it's impossible to run SIPS/SIPTLS through a nat'ed PBX.
Also, the combination of nat'ed client AND nat'ed PBXs is next to impossible to realize without dirty hacks which no firewall-admin will go along with.
SIP is fine for LAN or VPN applications. For the open internet and especially for mobile devices, IAX2 is the better choice.
Only people which need cleartext protocols because they don't understand binary protocols are against IAX2 ;).
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>IAX is a hack, a toy, a pseudo-protocol (instead of separating signalling and media as *any* modern VoIP protocol it forces both streams to go together).
LoL. First of all IAX2 was designed because of problems with SIP, especially in NAT environment. For an average user multiplexing works much better then separate RTP and signalling streams, and BTW, multiplexing is VERY common in telephony. Also IAX adds some features, including trunking support, dialplan exchange, etc, but it is a very different story. For me main benefits are:
1) It works via any NAT without ugly hacks like STUN.
2) In case of mulch-channel connection (> 1 stream) it saves bandwidth.
3) Better interoperability - too many ways in SIP to do encryption, pass DTMF, specification is not clear in many aspects, etc.
IAX2 is supported not only by Asterisk, but also by number of hardware and software products, including FreeSwitch. Some carriers providing IAX2 connection as well. So, from my point of view it would be great to see IAX2 support in CSIPSIMPLE. And for religious fanatics i would recommend just not to use it but not start this stupid flame.
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IAX is only producing less overhead on several calls .. unusual on cellphones.
calls on hold are not using a RTP channel as they are on hold on the Server.
a big disadvantage of IAX2 for mobile .. it doesnt work via TCP, UDP is a battery killer as allready known due the max refresh time of 60sec on some cellnetworks even down to 30sec.
In Comparation with TCP 15min ore more, typical is 60 min
i think to implement IAX2 for a Cellphone is not worth.
(as much i like IAX2)
IAX2 does include even the RTP stream, thats maybe one reason why there is never TCP forseen.
RTP via TCP will be a bandwith killer and using much more Batterie power while in Call. beside of the usual larger audiobuffer to overcome delay in the stream caused due retry on IP-Stack level, out of control of IAX2
on the end IAX2 was developed to multiplex streams into one, and share the channel TDMA like based .. which will never be realiable via TCP
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Presence could be provided via STOX or CUSAX, the two perspectives on Xmpp and SIP intersections. Jingle (xmpp) might well accomplish the NAT bliss claimed by IAX proponents who cannot also provide themselves entriprise configurations.
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Originally posted by: r3gis...@gmail.com
Hi,
That's not planned. IAX2 is really different from SIP. I'll try first to make
something suitable in SIP :).
As far as I know, pjsip, the native sip stack on which csipsimple is based, doesn't
support IAX2.
If the pjsip team decide to add something for IAX2, csipsimple will have this
improvement too.
Another solution would be to integrate another native opensource and portable stack,
but that's really far to be a priority.
I don't reject this request, but unless somebody else wants to contribute for that,
it is postponed for a long time.
Labels: -Type-Defect -Priority-Medium Type-Enhancement Priority-Low
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Originally posted by: kro...@gmail.com
I vote for this because of its smaller bandwidth requirements and better reliability
through firewalls. (Stun not required.) Huge advantage for android users on limited
data plans.
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Originally posted by: kro...@gmail.com
There is a library. Info provided by sipdroid user. Could win some market share. https://code.google.com/p/sipdroid/issues/detail?id=244
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Originally posted by: zdevel
voting for that.
IAX has great potential at mobiles, since it have some improvements in network part (better NAT handling, among others)
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Originally posted by: jpalka...@googlemail.com
vote for iax2 in android pls
as soon as I opened port 5060 to the world I got several dict attacks on sip login which basically DOS'ed my asterisk server, I had to firewall 5060 to just a few known IP's that will use my 'roid phone over wifi (will trial fail2ban to see if that alleviates the attacks)
I use an IAX client on my MacBook (Loudhush or JackenIAX) and have never seen an attack on 4569 on my asterisk server
also as mentioned IAX2 uses less b/w and avoids NAT issues
apart from this THANKS FOR CSipSimple
WORKS LOVEY ON ZTE BLADE (orange san francisco) under Gingerbread 2.3.3 Cyanogenmod7 (both RC1 and nightlies 1-15 so far)
audio routed correctly to earpiece and then loudspeaker when loudspeaking is chosen
sometimes audio quality lacks despite using codec 722 - trying KEEP AWAKE DURING CALL to see if that makes a difference
keep up the great work
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Originally posted by: bacon.li...@live.com
try IaxAgent by Johan Cloetens?
http://www.belgianwaves.com/
http://twitter.com/belgianwaves [I am not a twit]
http://www.appbrain.com/app/iaxagent-beta/com.bw.iax.ui
http://pbxinaflash.com/forum/showthread.php?t=7100
IAX handles NAT traversal orders of magnitude better than SIP
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Originally posted by: i...@aliax.net
IAX is a hack, a toy, a pseudo-protocol (instead of separating signalling and media as *any* modern VoIP protocol it forces both streams to go together).
IAX does not allow IM neither presence. IAX has several design issues and critical vulnerabilities and bugs in its main implementation (Asterisk).
IAX was created by Mark Spencer because he does not properly understand SIP protocol (given the old-fashion-PBX implemented in Asterisk that is not a surprise for me).
There is enough mechanisms in SIP for handling NAT, in client and server side. In the other side IAX requires that the media goes always to the central server which could be so far from both callers. This is a pain and un feasible in many environments, generating latency and so.
IAX is just supported by a few (very few) phones and just some PBX's (the anti-cool-DTMF-based Asterisk and FreeSwitch). Must we asume that CSipSimple is designed to "work with Asterisk"? Please not.
Please forget IAX. Just people with no knowledge of SIP is in favour of IAX (just because "it seems easier").
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Originally posted by: hbrau...@gmail.com
Whether you're for or against IAX2, matters not. The fact that SIP and IAX exsist, is sufficient enough to make them accessable to any user, anywhere.
CSipSimple is a great, stable program and if IAX can be added to it, then I say go for it.
Will save me running 2 apps and if it can all function in CSipSimple, that would only be a good thing..
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Originally posted by: r3gis...@gmail.com
Yep, actually I did a big effort on separating the SIP stack from the rest of the program. One of the next step will be to allow other sip stacks than pjsip (for example the stock gingerbread one or doubango, which is another great sip stack).
If this can be accomplished, I guess that it could be a good starting point to add other VoIP protocols support. The idea is to allow this kind of third party voip backend to be loaded as plugins apps, so would be a good compromise.
Android os is very well designed to allow to extends apps features with other plugins apps.
Well, as for this point, still open, still not in my higher priorities but contributions are welcome ;).
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Originally posted by: stefan.g...@gmail.com
@Comment 7:
IAX2 does support messaging and IIRC also presence. Combining the signaling and the audiostream into one connection has advantages in regards to protocol overhead. Additionally, an encrypted IAX2 stream is significantly harder to dissect than SIP because of the combination of signaling and audio.
One of the biggest disadvantages of SIP is that it's very hard to operate a nat'ed PBX and it's impossible to run SIPS/SIPTLS through a nat'ed PBX.
Also, the combination of nat'ed client AND nat'ed PBXs is next to impossible to realize without dirty hacks which no firewall-admin will go along with.
SIP is fine for LAN or VPN applications. For the open internet and especially for mobile devices, IAX2 is the better choice.
Only people which need cleartext protocols because they don't understand binary protocols are against IAX2 ;).
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Originally posted by: sammnet...@gmail.com
>IAX is a hack, a toy, a pseudo-protocol (instead of separating signalling and media as *any* modern VoIP protocol it forces both streams to go together).
LoL. First of all IAX2 was designed because of problems with SIP, especially in NAT environment. For an average user multiplexing works much better then separate RTP and signalling streams, and BTW, multiplexing is VERY common in telephony. Also IAX adds some features, including trunking support, dialplan exchange, etc, but it is a very different story. For me main benefits are:
1) It works via any NAT without ugly hacks like STUN.
2) In case of mulch-channel connection (> 1 stream) it saves bandwidth.
3) Better interoperability - too many ways in SIP to do encryption, pass DTMF, specification is not clear in many aspects, etc.
IAX2 is supported not only by Asterisk, but also by number of hardware and software products, including FreeSwitch. Some carriers providing IAX2 connection as well. So, from my point of view it would be great to see IAX2 support in CSIPSIMPLE. And for religious fanatics i would recommend just not to use it but not start this stupid flame.
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Originally posted by: marad...@gmail.com
Yes , We need IAX2 Support.
SIP is pain for nat and firewalls
When do you think IAX2 version release ?
Thanks
Mohamed
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Originally posted by: hb9...@gmail.com
IAX is only producing less overhead on several calls .. unusual on cellphones.
calls on hold are not using a RTP channel as they are on hold on the Server.
a big disadvantage of IAX2 for mobile .. it doesnt work via TCP, UDP is a battery killer as allready known due the max refresh time of 60sec on some cellnetworks even down to 30sec.
In Comparation with TCP 15min ore more, typical is 60 min
i think to implement IAX2 for a Cellphone is not worth.
(as much i like IAX2)
IAX2 does include even the RTP stream, thats maybe one reason why there is never TCP forseen.
RTP via TCP will be a bandwith killer and using much more Batterie power while in Call. beside of the usual larger audiobuffer to overcome delay in the stream caused due retry on IP-Stack level, out of control of IAX2
on the end IAX2 was developed to multiplex streams into one, and share the channel TDMA like based .. which will never be realiable via TCP
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Originally posted by: marad...@gmail.com
Hi Guys,
Did you create Iax2 for Android ?
Thanks
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Originally posted by: joshuacr...@gmail.com
Presence could be provided via STOX or CUSAX, the two perspectives on Xmpp and SIP intersections. Jingle (xmpp) might well accomplish the NAT bliss claimed by IAX proponents who cannot also provide themselves entriprise configurations.