Unable to build sip pvt data – Switching equipment congestion

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Asterisk Users 6 Comments

Hello list,

can anyone tell me what the following means (found in messages log) :

/[Nov 2 11:16:21] ERROR[18407] rtp.c: No RTP ports remaining. Can’t
setup media stream for this call.
[Nov 2 11:16:21] WARNING[18407] chan_sip.c: Unable to create RTP audio
session: Address already in use
[Nov 2 11:16:21] ERROR[18407] chan_sip.c: Unable to build sip pvt data
for ‘sipaccount7’ (Out of memory or socket error)
[Nov 2 11:16:21] WARNING[18407] app_dial.c: Unable to create channel of
type ‘SIP’ (cause 42 – Switching equipment congestion)/

Thank your for explaining the problems and a possible solution !

Greetingz,
Jonas.

6 thoughts on - Unable to build sip pvt data – Switching equipment congestion

  • You have set an insufficient range in rtp.conf. Asterisk uses 2 ports per
    call, but allocates 4 for transferring, etc, so when you set up a range of
    10001-10040 (for example) you are basically setting a range of 10 concurrent
    calls. Check rtp.conf and make the end range larger by 8 or 12 or whatever
    number of extra calls you’d like to see before you get this message again.

    [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jonas Kellens
    Sent: Wednesday, November 02, 2011 9:57 AM
    congestion

    Hello list,

    can anyone tell me what the following means (found in messages log) :

    [Nov 2 11:16:21] ERROR[18407] rtp.c: No RTP ports remaining. Can’t setup
    media stream for this call.
    [Nov 2 11:16:21] WARNING[18407] chan_sip.c: Unable to create RTP audio
    session: Address already in use
    [Nov 2 11:16:21] ERROR[18407] chan_sip.c: Unable to build sip pvt data for
    ‘sipaccount7’ (Out of memory or socket error)
    [Nov 2 11:16:21] WARNING[18407] app_dial.c: Unable to create channel of
    type ‘SIP’ (cause 42 – Switching equipment congestion)

    Thank your for explaining the problems and a possible solution !

    Greetingz,
    Jonas.

  • Hello,

    thank you for your answer…

    Current range (rtp.conf) : 11500 – 11650

    Current calls : 20 à 25

    Is this not sufficient ??

    Jonas.

  • “Number of wished concurrent calls” times 4 = “Number of ports you’ll
    need to setup in rtp.conf” 😉

  • 150/4 = 37.5. maybe your sip peer has a conflicting range?

    [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jonas Kellens
    Sent: Wednesday, November 02, 2011 10:06 AM
    equipment congestion

    Hello,

    thank you for your answer…

    Current range (rtp.conf) : 11500 – 11650

    Current calls : 20 à 25

    Is this not sufficient ??

    Jonas.

    You have set an insufficient range in rtp.conf. Asterisk uses 2 ports per
    call, but allocates 4 for transferring, etc, so when you set up a range of
    10001-10040 (for example) you are basically setting a range of 10 concurrent
    calls. Check rtp.conf and make the end range larger by 8 or 12 or whatever
    number of extra calls you’d like to see before you get this message again.

    [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jonas Kellens
    Sent: Wednesday, November 02, 2011 9:57 AM
    congestion

    Hello list,

    can anyone tell me what the following means (found in messages log) :

    [Nov 2 11:16:21] ERROR[18407] rtp.c: No RTP ports remaining. Can’t setup
    media stream for this call.
    [Nov 2 11:16:21] WARNING[18407] chan_sip.c: Unable to create RTP audio
    session: Address already in use
    [Nov 2 11:16:21] ERROR[18407] chan_sip.c: Unable to build sip pvt data for
    ‘sipaccount7’ (Out of memory or socket error)
    [Nov 2 11:16:21] WARNING[18407] app_dial.c: Unable to create channel of
    type ‘SIP’ (cause 42 – Switching equipment congestion)

    Thank your for explaining the problems and a possible solution !

    Greetingz,
    Jonas.

  • Where do I set this range in my peer definition ? I don’t think there is
    such a parameter in sip.conf

    To be perfectly clear, how many RTP-ports are needed in the below
    situation :

    – 1 peer answers this incoming call

    My thought : 2 RTP for incoming channel, 2 RTP for channel to SIP peer
    (and the other peers don’t matter)

    Am I correct ?

    Or is there a need for a channel to every peer that is “ringing” ?

  • As I understand it, the scenario you describe would only use 2 channels (I
    don’t think the RTP channel gets established until connection; I could be
    wrong about this as Asterisk might pre-reserve the channels for early media,
    etc.) – do keep in mind however that although you mention 2 and 2 (1 each
    for incoming channel and ringing/answered), you have actually “blocked out”
    8 channels as the range of 4 per call is used.

    For example

    SIP/100 calls SIP/101

    Call from SIP/100 uses 11501 and 11502 and 11503 and 11504 are reserved for
    transfer, etc.

    SIP/101 answer uses 11505 and 11506 and reserves 11508/11509 for its next
    steps.

    I would suggest doing a netstat –anp while you are ringing your 10 peers to
    see which RTP ports are in use.

    [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jonas Kellens
    Sent: Wednesday, November 02, 2011 10:20 AM
    equipment congestion

    150/4 = 37.5. maybe your sip peer has a conflicting range?

    Where do I set this range in my peer definition ? I don’t think there is
    such a parameter in sip.conf

    To be perfectly clear, how many RTP-ports are needed in the below situation
    :

    – 1 peer answers this incoming call

    My thought : 2 RTP for incoming channel, 2 RTP for channel to SIP peer
    (and the other peers don’t matter)

    Am I correct ?

    Or is there a need for a channel to every peer that is “ringing” ?

    [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jonas Kellens
    Sent: Wednesday, November 02, 2011 10:06 AM
    equipment congestion

    Hello,

    thank you for your answer…

    Current range (rtp.conf) : 11500 – 11650

    Current calls : 20 à 25

    Is this not sufficient ??

    Jonas.

    You have set an insufficient range in rtp.conf. Asterisk uses 2 ports per
    call, but allocates 4 for transferring, etc, so when you set up a range of
    10001-10040 (for example) you are basically setting a range of 10 concurrent
    calls. Check rtp.conf and make the end range larger by 8 or 12 or whatever
    number of extra calls you’d like to see before you get this message again.

    [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jonas Kellens
    Sent: Wednesday, November 02, 2011 9:57 AM
    congestion

    Hello list,

    can anyone tell me what the following means (found in messages log) :

    [Nov 2 11:16:21] ERROR[18407] rtp.c: No RTP ports remaining. Can’t setup
    media stream for this call.
    [Nov 2 11:16:21] WARNING[18407] chan_sip.c: Unable to create RTP audio
    session: Address already in use
    [Nov 2 11:16:21] ERROR[18407] chan_sip.c: Unable to build sip pvt data for
    ‘sipaccount7’ (Out of memory or socket error)
    [Nov 2 11:16:21] WARNING[18407] app_dial.c: Unable to create channel of
    type ‘SIP’ (cause 42 – Switching equipment congestion)

    Thank your for explaining the problems and a possible solution !

    Greetingz,
    Jonas.